Analysis of Channel Estimation Methods For

Analysis of channel estimation methods for
OFDMA
DANIEL LARSSON
Master of Science Thesis

Stockholm, Sweden 2006-12-19
XR-EE-KT 2006:011
2
Abstract
This master thesis analyses different channel estimation techniques in an OFDMA
(Orthogonal Frequency Division Multiple Access) system with parameter settings
from IEEE 802.16e and WiMAX (Worldwide Interoperability for Microwaves
Access). IEEE 802.16e is one of the standards that create the foundation for
WiMAX. OFDM (Orthogonal Frequency Division Multiplexing) is a transmission
technique that is based on many orthogonal carriers that transmits simultaneously.
All the channel estimators studied uses pilot symbols to estimate the radio channel.
The analysis of the channel estimators has been conducted using a Matlab®
simulator implemented for this master thesis. The simulator evaluated the different
channel estimators for different frame allocations and at different relative speeds.
Two major conclusions were reached. The first is that the best method to estimate
the radio channel is to estimate its taps and the second is that the choice of
interpolation method affects the estimation considerably.
3
Acknowledgements
First of all, I want to thank Lars Lindbom for his advice and time during the work
with the master thesis. I also want to thank TietoEnator Baseband for giving me the
opportunity to perform my master thesis at their office. Finally I also want to thank
my advisor at KTH Svante Bergman and my examiner Erik Larsson.
5
Table of Contents
CHAPTER 1 INTRODUCTION ................................................................. 17

1.1 OFDM.............................................................................................. 17
1.2 WiMAX ........................................................................................... 17
1.3 Objective .......................................................................................... 18
1.4 Previous work................................................................................... 18
1.5 Outline ............................................................................................. 18
CHAPTER 2 SYSTEM MODEL................................................................. 20

2.1 Basic principles of OFDM ................................................................ 20
2.1.1 Cyclic prefix ................................................................................ 22
2.1.2 OFDM in systems ........................................................................ 23
2.1.3 OFDMA....................................................................................... 24
2.2 IEEE 802.16e ................................................................................... 25
2.2.1 Frame structure ............................................................................ 26
2.2.2 Pilot generation ............................................................................ 27
2.3 Simulator.......................................................................................... 28
2.3.1 Transmission model ..................................................................... 28
2.3.2 Channel model ............................................................................. 30
2.3.3 Signal to noise ratio...................................................................... 34
2.3.4 Simulation settings ....................................................................... 35
CHAPTER 3 CHANNEL ESTIMATION ................................................... 39

3.1 1D estimator ..................................................................................... 39
3.1.1 Least squares channel estimator.................................................... 39
3.1.2 FIR interpolation algorithm .......................................................... 41
3.1.3 LMMSE....................................................................................... 44
3.2 2D estimator ..................................................................................... 46
3.2.1 2D MMSE channel estimator........................................................ 46
3.2.2 Low complexity 2D channel estimator.......................................... 47
3.3 Adaptive estimator............................................................................ 49
3.3.1 Kalman estimator ......................................................................... 49
3.3.2 The low-complexity Kalman estimator ......................................... 50
3.4 Interpolation ..................................................................................... 52
3.4.1 Interpolation method 1 in frequency ............................................. 53
3.4.4 Interpolation method 1 in time...................................................... 56
3.5 Summary .......................................................................................... 56
3.5.1 1D estimators ............................................................................... 56
3.5.2 2D estimation............................................................................... 57
3.5.3 Adaptive estimation...................................................................... 57
CHAPTER 4 SIMULATIONS..................................................................... 59
4.1 The Test cases .................................................................................. 59
7
4.1.1 Specification case 1 ..................................................................... 60
4.2 Tuning of channel estimators............................................................ 62
4.3 Case 1 .............................................................................................. 64
4.3.1 Simulation case 1......................................................................... 64
4.3.2 Conclusion case 1 ........................................................................ 68
4.4 Case 2 .............................................................................................. 68
4.4.1 Simulation case 2......................................................................... 68
4.4.2 Conclusions case 2....................................................................... 72
4.5 Case 3 .............................................................................................. 72
4.5.1 Simulation case 3......................................................................... 72
4.5.2 Conclusion case 3 ........................................................................ 76
4.6 General Conclusions......................................................................... 76
CHAPTER 5 CONCLUSIONS AND FUTURE WORK ............................ 81
REFERENCES................................................................................................. 84
8
List of Tables
Table 2.1: Information about uplink simulation setup .......................................... 35
Table 4.1: Settings for the simulations................................................................. 60
Table 4.2: Settings for case 1 .............................................................................. 60
Table 4.5: Properties that channel estimators utilize............................................. 77
Table 4.6: Case 1 profile..................................................................................... 78
10
List of Figures
Figure 2.1: OFDM spectra .................................................................................. 21
Figure 2.2: Remove ISI through guard interval.................................................... 22
Figure 2.3: Cyclic Prefix..................................................................................... 22
Figure 2.4: Example of OFDM signal pattern ...................................................... 24
Figure 2.5: Example of an OFDMA communication............................................ 24
Figure 2.6: Example of a frame structure for IEEE 802.16e in TDD mode [1]...... 26
Figure 2.7: Uplink Tile ....................................................................................... 27
Figure 2.8: PRBS generator [1]........................................................................... 27
Figure 2.9: OFDM transmission model ............................................................... 28
Figure 2.10: Layout of an uplink frame ............................................................... 36
Figure 2.11: Tile positioning in frame ................................................................. 36
Figure 3.1: LS implementation............................................................................ 41
Figure 3.2: FIR interpolation algorithm implementation ...................................... 43
Figure 3.3: LMMSE implementation................................................................... 46
Figure 3.4: Low complexity 2D implementation.................................................. 49
Figure 3.5: Low-complexity Kalman implementation .......................................... 52
Figure 4.1: Simulation case 1.............................................................................. 60
Figure 4.4: High speed simulation parameter b.................................................... 63
Figure 4.5: Mid speed simulation parameter b ..................................................... 63
Figure 4.6: No speed simulation parameter b....................................................... 64
Figure 4.7: Case 1 no speed simulation part 1...................................................... 65
Figure 4.8: Case 1 no speed simulation part 2...................................................... 65
Figure 4.9: Case 1 mid speed simulation part 1.................................................... 66
Figure 4.10: Case 1 mid speed simulation part 2.................................................. 66
Figure 4.11: Case 1 high speed simulation part 1................................................. 67
Figure 4.13: Case 2 no speed simulation part 1.................................................... 69
12
List of Abbreviations
1D 1 Dimension
2D 2 Dimensions
A
AIC Akaike’s Information Criterion
AWGN Additive White Gaussian Noise
B
BPSK Bit Phase Shift Keying
BS Base Station
C
CP Cyclic Prefix
D
DEMUX Demultiplexer
DFT Discrete Fourier Transform
DL Downlink
DVB-T Digital Video Broadcasting standard - Terrestrial
F
FDD Frequency Division Duplexing
FDM Frequency Division Multiplexing
FFT Fast Fourier Transform
FIR Finite Impulse Response
G
GWSSUS Gaussian Wide-Sense Stationary Uncorrelated Scattering
I
ICI Inter Carrier Interference
IDFT Inverse Discrete Fourier Transform
IEEE Institute of Electrical and Electronics Engineering
IFFT Inverse Fast Fourier Transform
ISI Inter Symbol Interference
L
LMMSE Least Minimum Mean Square Error
LOS Line Of Sight
LS Least Squares
M
MIMO Multiple Input Multiple Output
ML Maximum Likelihood
MSB Most Significant Bit
14
LIST OF ABBREVIATIONS 15
MUX Multiplexer
N
NLOS Non Line Of Sight
O
OFDM Orthogonal Frequency Division Multiplexing
OFDMA Orthogonal Frequency Division Multiple Access
P
PRBS Pseudo Random Bit Sequence
PUSC Partially Used Subcarrier
Q
QAM Quadrature Amplitude Modulation
QPSK Quadrature Phase Shift Keying
R
RTG Receive Transition Gap
S
SISO Single Input Single Output
SVD Singular Value Decomposition
SNR Signal to Noise Ratio
SOFDMA Scalable Orthogonal Frequency Division Multiple Access
T
TDD Time Division Duplex
TTG Transmit Transition Gap
U
UL Uplink
W
WiMAX Worldwide Interoperability for Microwaves Access
WLAN Wireless Local Area Network
X
X-DSL X-Digital Subscriber Line
XOR Exclusive Or
Chapter 1
Introduction
This master thesis analyses different channel estimation techniques for an OFDMA
(Orthogonal Frequency Division Multiple Access) system with parameters from the
standards IEEE 802.16e and WiMAX. This chapter gives some background
information on OFDM and WiMAX, defines the problem that is investigated,
describes previous work in the field and gives an outline of the report.
1.1 OFDM
Orthogonal frequency division multiplexing (OFDM) is a transmission technique
that is built-up by many orthogonal carriers that transmits simultaneously. The main
idea behind OFDM is that a signal with a long symbol duration time is less
sensitive to multipath fading, than a signal with a short symbol time. Hence, a gain
in performance can be achieved by sending several parallel symbols with a long
symbol time than sending them in a series with a shorter symbol time. The basic
technique of OFDM has been known for around 40 years [12], but it is not since
recently that OFDM has become widely used. Products that use OFDM are for
example WiMAX, WLAN (Wireless Local Area Network) 802.11, x-DSL (x-
Digital Subscriber Line) and DVT (Digital Video Broadcasting).
1.2 WiMAX
WiMAX (Worldwide Interoperability for Microwaves Access) is a standard-based
wireless technology. WiMAX is primarily built upon the two standards IEEE
802.16-2004 and 802.16e (802.16-2005). The WiMAX Forum is the organisation
that specifies WiMAX. Since the two IEEE standards are too extensive to achieve
interoperability among vendors, the WiMAX Forum has limited the specifications.
The primary focus on IEEE 802.16-2004 is on fixed point-to-point communication,

both for LOS (Line Of Sight) and NLOS (Non Line Of Sight) scenarios. The focus
in IEEE 802.16e is on a multi-user communication and mobility. In this master
thesis general parameter settings are used from IEEE 802.16e and therefore IEEE
802.16-2004 will not be discussed in any depth.
17
18 CHAPTER 1. INTRODUCTION
1.3 Objective
The purpose of this master thesis is to study different channel estimation techniques
for an OFDMA system with its general parameters from IEEE 802.16e. The
standard covers both SISO (Single Input Single Output) and MIMO (Multiple Input
Multiple Output) communication system. In this thesis only the SISO case will be
considered. The main feature in IEEE 802.16e is the mobility and mulit-user
support. In this master thesis, the mobility aspect is considered, while the multi-user
aspect is only considered to some extent. In some parts of the thesis specific
WiMAX parameters are used, in these cases it will be clearly stated.
The channel estimators that are analysed can generally be split up into two parts,
estimation and interpolation. The main focus is on estimation. However, a few
interpolation techniques are also studied in order to compare the impact
interpolation has on the performance.
1.4 Previous work

A general description of OFDM modulation is given in [16] and also in [5]. A
description of the IEEE 802.16e is given in the standard documents [1] and [2]. An
overview of the main parts in a WiMAX system can be found in [15].
Studies that to some extent compare different channel estimation and interpolation
techniques for a SISO OFDM systems have been carried out in [3], [5] and [8]. The
channel estimation techniques used in this master thesis are modifications of the
estimators described in [3], [4], [5], [7] and [8].
1.5 Outline
The outline of this master thesis is as follows.
• The System model chapter, Chapter 2, describes the theory behind OFDM,
IEEE 802.16e and the simulator.
• The channel estimation chapter, Chapter 3, describes the signal model used
for the simulations. The chapter explains the different channel estimators
and interpolation techniques covered by this thesis.
• In the Simulation chapter, Chapter 4, the different simulation test cases are
defined and the results from the simulations are presented. The results are
interpreted and conclusions are presented.
• The last chapter, Chapter 5, summarises the conclusions from the
simulations and also looks ahead on the subjects that require further study.
Chapter 2
System Model
A system model is required for the simulations. This chapter describes the
theoretical background of the simulator as well as the specific parameters used in
the simulator.
2.1 Basic principles of OFDM

OFDM consists of many orthogonal carriers; each carrier is called a subcarrier or
tone, depending on literature. Usually some sort of QAM (Quadrature Amplitude
Modulation) is used to modulate the symbols that are transmitted on the subcarriers.
When a symbol is transmitted on a subcarrier its transmission time is extended N
times. However there is no reduction in symbol rate since there are N subcarriers
transmitting N symbols during the time interval NT, where T is the original symbol
rate. All symbols that are transmitted during the time interval NT forms an OFDM
symbol.
The advantage with OFDM systems is the ability to completely remove ISI (Inter
Symbol Interference) between OFDM symbols. The ISI is usually removed by
adding a cyclic prefix to the OFDM symbol before transmitting it. How this is done
is described in more depth in section 2.1.1. A disadvantage with an OFDM system
is that usually the subcarriers will not be orthogonal when received at the receiver
due to Doppler shift and different frequencies in the local oscillators at the
transmitter the receiver. Hence, this frequency offset has to be estimated.
Since the ISI can be removed; each subcarrier will experience only a flat fading
channel. This statement would also hold for an FDM system (Frequency division
multiplexing), but in FDM there are guard bands between each carrier. Where as in
OFDM there are no guard bands, it is even so that the different subcarriers share
some of the spectra as seen in Figure 2.1.
20
2.1. BASIC PRINCIPLES OF OFDM 21
Figure 2.1: OFDM spectra
In Figure 2.1 there are five subcarriers, each subcarrier overlap in some part all the
other subcarriers. However a receiver can still extract the symbols sent on each
subcarrier since the subcarriers are orthogonal, i.e. at energy maximum of each
subcarrier no other subcarrier contributes with any energy.
A baseband OFDM system model can be expressed in the following manner, where
each subcarrier is of the form
e j 2πf k n , (2.1)
where n is a discrete time variable and f k is the carrier frequency for subcarrier
k . The subcarriers frequency f k is defined as
k
fk = , (2.2)
NT
where N is the number of subcarriers and T is the original sample time of the
transmitted symbol. The baseband model of an OFDM system in discrete time is
N −1
x( n ) = ∑ c(k )e
k =0
j 2πf k n
, 0 ≤ n ≤ N −1 , (2.3)
22 CHAPTER 2. SYSTEM MODEL
where c(k ) is the transmitted symbol on subcarrier k and x(n) is an OFDM

symbol. Interestingly, expression (2.3) is the IDFT (Inverse Discrete Fourier
Transform). This implies that the demodulation of the signal x(n) can be done
using a DFT (Discrete Fourier Transform). In reality an FFT (Fast Fourier
Transform) is used for demodulation because of the gain in computational speed
compared to the DFT convolution (2.3) and an IFFT (Inverse Fast Fourier
Transform) is used to modulate the signal.
2.1.1 Cyclic prefix
As previously mentioned an OFDM system has the ability to completely remove ISI
between two OFDM symbols. A simple solution, which would remove the ISI, is to
simply insert a guard interval between the OFDM symbols, i.e. simply wait for all
the multipaths reflections of the transmitted OFDM symbol to fade out before
transmitting another OFDM symbol, as in Figure 2.2.
Figure 2.2: Remove ISI through guard interval
There is however a problem using guard intervals to remove the ISI, which has do
with the property of the DFT. The DFT is cyclic, so if the received OFDM symbol
is not cyclic, it will cause ICI (Inter Carrier Interference) between the subcarriers.
The solution is to add a cyclic extension to the OFDM symbol before transmitting
it. The cyclic extension that is added before the transmission is simply the end part
of the OFDM symbol that has been copied and transmitted before the OFDM
symbol as in Figure 2.3. The cyclic extension is referred to as CP (Cyclic Prefix).
Figure 2.3: Cyclic Prefix
The length of the CP is set to at least to the maximum length of multipath delay of
the radio channel, i.e. to at least the same number of taps as the channel. The
corresponding baseband model for an OFDM system with cyclic prefix is
N −1
x( n ) = ∑ c(k )e
k =0
j 2πf k n
, − CP ≤ n ≤ N − 1 , (2.4)
where CP denotes the length of the cyclic prefix. The drawback of the cyclic
prefix is that it does not carry any new data, which lowers the transmitted energy
per information bit and thereby decreasing the SNR.
2.1. BASIC PRINCIPLES OF OFDM 23
2.1.2 OFDM in systems
We begin this subsection by describing some aspects that has to be considered

when implementing an OFDM system and continues with aspects that are more
central for the master thesis. In most mobile communication systems, not only
OFDM systems, some sort of channel coding is used to lower the bit error by
creating redundancy. This is also the case for most of OFDM systems, since the
overhead of the channel coding is usually far smaller than having to retransmit the
faulty information.
There is often a need to communicate in both directions, i.e. between BS (base

station) and the terminal and vice versa. The two main methods for this in OFDM
are FDD (Frequency Division Duplex) and TDD (Time Division Duplex). In an
FDD system communications in downlink (from base station to terminal) and
uplink (from terminal to base station) are separated in two different frequency
bands. In a TDD system, the communication in downlink and uplink is on the same
frequency bands but is split up in time, e.g. first the transmission in downlink and
then in uplink.
As previously mentioned some sort of channel coding is often used to lower the bit
error rate. But an OFDM system still requires a frequency offset estimator and a
channel estimator to achieve acceptable performance. The frequency offset
estimator is required to counter the affect of offsets in frequencies between the local
oscillators at the transmitter and the receiver that otherwise would destroy the
orthogonality between the subcarriers. If the subcarriers are not orthogonal, they
will cause ICI and then the sent information will be very difficult to reconstruct.
Since the focus in this master thesis report is on channel estimation, frequency
offset estimation is assumed to be ideal. In the case of channel estimation it is here
primarily done in the frequency dimension, i.e. after the demodulation of the
OFDM signal. Due to no ISI, each subcarrier is only affected by a multiplicative
complex valued scalar in the frequency domain, which is to be estimated. These
scalars are channel dependent and may differ over time and also over different
subcarriers.
The most preferred and used method to estimate the channel and the offset in
frequency is to use pilot symbols. Pilot symbols are symbols that are known to the
transmitter and receiver in advance. The basic idea with pilot symbols is that there
is a strong correlation between the pilot symbol fading and the fading of
information data symbols that are sent close to the pilot symbol in time and
subcarrier.
The pilot symbols and the information data symbols are typically placed in some
kind of pattern on the different subcarriers and over time, as seen in Figure 2.4
Figure 2.4: Example of OFDM signal pattern
In Figure 2.4 a new definition is also presented, the null symbols. The null symbols
are typically sent on the null subcarriers, which simply are subcarriers that do not
have any transmission and hence the name. Some OFDM communication systems
may have null subcarriers at the lower and high subcarriers to act as a guard so that
the OFDM communication system does not interfere with other devices that are
communicating close in frequency. Also some frequencies are not used as they are
used on the cell-edge of other users in other cells. A vertical line in Figure 2.4
corresponds to an OFDM symbol, in the example figure there are eight OFDM
symbols.
2.1.3 OFDMA
OFDMA (Orthogonal Frequency Division Multiple Access) is a multi-user version

of OFDM, and all that were previously mentioned about OFDM also holds for
OFDMA. Each user in an OFDMA system is usually given certain subcarriers
during a certain time to communicate. Figure 2.5 show an example of OFDMA
communication.
Figure 2.5: Example of an OFDMA communication

2.2. IEEE 802.16E 25
One of the major problems with an OFDMA system is to synchronize the uplink
transmission, because every user has to transmit its frame so that they avoid
interfering the other users. For example as in Figure 2.5, if User 2 transmits too
early it will disturb some of the User 1 transmission and if it transmits too late it
will disturb User 4. On the downlink side this problem will not arise since the signal
originates from a single source.
2.2 IEEE 802.16e

IEEE 802.16e uses SOFDMA (Scalable Orthogonal Frequency Division Multiple
Access) as transmission technique. SOFDMA is an OFDMA version where the
bandwidth is scalable; in 802.16e it is scalable between 1.25 to 20 MHz. The
scalability is achieved by changing the FFT size, while keeping fixed subcarrier
spacing [15]. The communication in 802.16e is based on a frame structure that is
described in more detail in section 2.2.1.
Mobile WiMAX is intended for the 2.3 GHz, 2.5 GHz and 3.5 GHz [15] spectra.
The system is defined so that the user can travel at speeds between 0-120 km/h. The
theoretical upper limit for the bit rate in WiMAX, given a bandwidth of 10 MHz, is
31 Mbps in downlink and 23 Mbps in uplink [15]. The base stations have a typical
coverage up to an 8 km radius in a NLOS environment [17]. One of the more
interesting features of 802.16e is that it supports MIMO (Multiple Input Multiple
Output) devices, although this is not considered in this master thesis.
The market that 802.16e is focused on is typical city area network. But in the future
wireless devices such as laptops and cell phones will probably be able to access
different types of wireless networks depending on availability.
2.2.1 Frame structure
An example of the frame structure in TDD mode in IEEE 802.16e (802.16-2005) is

presented in Figure 2.6.
Figure 2.6: Example of a frame structure for IEEE 802.16e in TDD mode [1]
An 802.16e TDD frame is built up by one downlink (DL) subframe and one uplink
(UL) subframe. In this thesis, only the uplink subframe structures will be
considered. In the example in Figure 2.6, the uplink frame is divided into five
different UL burst areas, which implies that are five different terminals that are
communicating with the BS (base station). In general a terminal is given a number
of subcarriers both in the frequency and time dimension based on how much
throughput the device require and also how many other terminals that are connected
to the BS.
To separate the downlink frame from the uplink frame, guard zones are inserted as
seen in Figure 2.6. In the case between the downlink and uplink frame the guard
zone is TTG (Transmit Transition Gap) and between the end of the frame and the
next frame the guard zone is RTG (Receive Transition Gap).
The smallest structure in the uplink frame is a tile. In SISO PUSC (Partially Used
Sub-Carrier) mode, a tile is built up according to Figure 2.7. PUSC is an uplink
mode that must be supported by all WiMAX devices.
2.2. IEEE 802.16E 27
Figure 2.7: Uplink Tile
The data symbols can be modulated with QPSK, 16QAM or 64QAM. One terminal
uses only one of the previous mentioned modulations based on the SNR value.
However a whole uplink frame can contain different modulations since different
terminals may use different modulations. A few different outer coding techniques
for error protection are supported by the standard. However, the focus in the thesis
is on channel estimation and the impact it has on performance, no outer codes are
therefore considered.
The pilot symbols are generated using a PRBS (Pseudo Random Bit Sequence),
which is described in more detail in section 2.2.2.
2.2.2 Pilot generation
To generate the pilot symbols a PRBS (Pseudo Random Bit Sequence) generator is
used, see Figure 2.8. The same PRBS generator is defined in 802.16 [2] and
802.16e [1].
Figure 2.8: PRBS generator [1]
The PRBS is initiated by first setting the 11 bits register to a value based on the
frame number and a few other parameters that are presented in [1] and [2]. For
every new pilot symbol that should be generated, the eleventh bit ( wk ) is taken
from the register, where k is the subcarrier index. The pilot symbol bk is then
calculated as
1 
bk = 2 − wk  . (2.5)
2 
At the same time the ninth and eleventh bit from the register is XOR:ed and the
result are placed at the MSB position and finally the register is shifted to the right.
The main reason for using a PRBS to generate the pilot symbols is that it gives a
sufficient random sequence given a good initiation value. Another aspect is that the
PRBS is rather easy to implement in software and particularly on a chip.
2.3 Simulator
A simulator that accurately models the transmission on radio channels would be too
complex to implement. Commonly, one applies approximations and simplifications
that reduce the computational complexity while preserving the most significant
aspects of the actual system. The simulator developed herein to test the different
channel estimation techniques has been developed using Matlab®. This chapter will
present the transmission model, the channel model and finally the IEEE 802.16e
parameters that were used in configuration of the simulator.
2.3.1 Transmission model
Figure 2.9: OFDM transmission model
A typical transmission model for an OFDM system is as shown in Figure 2.9. The
transmission model is adopted by the simulator. The model describes how an
OFDM symbol is processed, where xi,j corresponds to a transmitted symbol at
subcarrier i at time j and yi,j is the received symbol at subcarrier i at time j .
Simulations are carried out using a 10 MHz bandwidth, which corresponds to a
system with 1024 subcarriers in 802.16e, i.e. the length of the IDFT and DFT is
1024.
The cyclic prefix (CP) in the Figure 2.9 represents the end part of the result from
the N-point long IDFT data sequence and is added to the beginning of the data
sequence. In the simulations it is assumed that the cyclic prefix is chosen large
enough so that there is no ISI and the orthogonality is preserved. The radio channels
impulse response is denoted by h(τ,t), where h(τ,t) is the impulse response with
2.3. SIMULATOR 29
respect to τ at the time instance t. In the simulations Jake’s channel model [6] is
used, which is described in more detail in section 2.3.2.
The simulator implements the transmission model in Figure 2.9. The transmission
model can also be described in the following algebraic form, assuming that N
symbols have been serial to parallel converted. To derive the signal model used in
the simulations requires the following basic definitions, given that there are N
subcarriers in one OFDM symbol
 d ( 0) L 0 

D= M O M , (2.6)

 0 L d ( N − 1)
 w00N L wN0( N −1) 

 
WN =  M O M , (2.7)
w( N −1)0 L w( N −1)( N −1) 
 N N 
 h (0 ) 
 
h(1) 
h= , (2.8)
 M 
 
h( N − 1)
 e~(0) 
 ~ 
~e =  e (1)  , (2.9)
 M 
~ 
e ( N − 1)
where WN is the N-by-N DFT-matrix, D with the sent symbol d on its diagonal,
h is the channel impulse response and ~e is the DFT of the noise e that is AWGN
with the variance σ n2 .
In the simulations it is assumed that the cyclic prefix is chosen longer than the
channel impulse response, thus removing ISI. The received signal can then be
written
r (n ) = IDFT (d (n )) ⊗ h(n ) + e(n) (2.10)
Assuming that the cyclic prefix has been removed, in (2.10), ⊗ denotes cyclic
convolution, d (n ) is the sent data symbol before OFDM modulation, h(n) is the
channels impulse response and e(n ) is AWGN. The OFDM receiver demodulates
the received OFDM symbol with a DFT. Since a DFT of a cyclic convolution of the
two signals equals the product, the DFT of the received signal r (n ) is
y (n ) = DFT (IDFT (d (n )) ⊗ h(n ) + e(n )) =

, (2.11)
d (n )DFT (h(n )) + e~ (n )
where ~e (n) is the DFT (e(n )) . The demodulated received signal y (n ) on a matrix
formulation becomes
y = DWNh + ~e , (2.12)
where D is the sent symbols defined in expression (2.6), WN h is the DFT of the
channels impulse response and ~e is the DFT of AWGN with variance σ 2 . The n
channel estimation is performed after demodulation of the OFDM symbol based on
the received vector y .
2.3.2 Channel model
In order to do simulations as close to the reality as possible, it is important to have a

good channel model. The channel model used in this simulation has to support both
an urban area and mobility of the client terminals.
A main property of a channel is if it is time-variant or time-invariant. Time-

invariant channels are constant over time and are relatively easy to estimate since it
only requires one estimate at the beginning of the reception of the signal. A time-
variant channel varies over time and is much harder to estimate since it requires
continuous estimation. Depending how fast the channel varies between two
successive symbols it is said to be either fast or slow fading. Often slow fading
occurs due large objects that move relative to the radio device (for example
mountains). Fast fading occurs due to multipath components of the signal that mix
up either constructively or destructively. Hence as the terminal moves on the order
of half a wavelength a constructive superposition becomes destructive and vice
versa.
In many cases a receiver receives more than one copy of the transmitted signal due
to multiple paths between the transmitter and the receiver, i.e. a multipath channel.
In such a case the channel is frequency selective, i.e. the channel varies in
frequency. The other case is when receiver only receives one copy of the
transmitted signal, which results in a flat frequency spectrum.
In general the receiver and transmitter do not have a direct line of sight between
them. Instead the radio signal is reflected by objects before it reaches the receiver.
Such a channel model is described in [13] and can be modelled as
c(t ) = c r (t ) + jci (t ) , (2.13)

2.3. SIMULATOR 31
where cr (t ) and ci (t ) are, stationary, statistically independent and real-valued

Gaussian processes. Note, that cr (t ) and ci (t ) are typical not white processes, but
instead coloured Gaussian. It is possible to rewrite c (t ) on polar form as
c(t ) = a(t )e jφ (t )
, (2.14)
where
a(t ) = cr2 (t ) + ci2 (t ) ,

ci (t ) (2.15)
φ (t ) = tan −1 .
cr (t )
If there are enough different paths the central limits theorem will model the channel
as a Rayleigh distributed channel, i.e. cr (t ) and ci (t ) are Gaussian with zero-mean.
The amplitude a(t ) is then Rayleigh distributed and φ (t ) becomes uniformly
distributed between (0,2π ) . The Rayleigh distribution is described by its PDF
− a2
p(a ) =
a 2σ 2
e a≥0, (2.16)
σ 2
[ ] [ ]
where σ 2 = E c r2 = E ci2 and p(a ) = 0 for a < 0 .
If there exist a line of sight component the channel can be modelled as a Riciean
distribution. Since in this thesis it is assumed that there is no line of sight between
the transmitter and the receiver, no channel model for that case is studied.
When the receiver and transmitter are moving relative to each other it causes the
carrier frequency at the receiver to shift, this is known has maximum Doppler shift
v d max f c
fd = , (2.17)
c
where f d is the Doppler shift, v is the relative velocity, c is the propagation speed
of the signal (speed of light) and f c is the carrier frequency. This connects to what
was previously mentioned about fast and slow fading since the receiver and
transmitter moving relative to each other is the main cause of fading.
The simulator uses Jake's channel model [6] to generate fading channels. The
impulse response of the channel is described by
M
h (t , τ ) = ∑α
k =1
k (t )δ (τ −τ k ) , (2.18)
where τ k is the delay of path k and α k (t ) is the corresponding complex amplitude.

The channel model that is assumed does not take into account that the number of
delay paths M and the delays τ k changes over time. Every α k (t ) is a complex
Gaussian zero mean and can be described as a function of an angle θ k (t ) and the
distance d k (t ) between the receiver and the transmitter as follows
M
α k (t ) = ∑e π
k =1
2 d k (t ) cos(θ k ( t ))
(2.19)
The autocorrelation of the different transmission paths α k (t ) is a zero order Bessel

function that is depending on the time difference τ and the Doppler shift,
J 0 (2πf dτ ) . The factor that controls how much the channel varies between to
successive symbols is f d τ .
A channel model also has to have some kind of restriction of how the power is
distributed in time. The two most common types of power delay distribution are
uniformly and exponential. The uniformly distribution implies that the power of the
channel response is uniformly distributed over a certain time. The exponential
power delay profile, which is defined in [4] on the other hand, has its power
distributed according to
−τ k
τ rms
θ (k ) = Ce , (2.20)
where τ k is the delay of path k, C is an arbitrary constant and τ rms is the root-
mean-square delay spread of the channel. The delays τ k are all uniformly
distributed over the delay time of the channel. Taking this in to account the channel
can now be defined as
M −τ k
h(t ,τ ) = ∑
k =1
α k (t ) Ce τ rms
δ (τ − τ k ) (2.21)
In the OFDM case different subcarriers correspond to different frequencies. Hence,

it is of interest to derive the time varying transfer function of the channel, which
yields the following expression assuming that an exponential power delay profile is
used with C set so that the average power is normalized to one.
∞ M −τ k
H (t , f ) = ∫ h(t ,τ )e − j 2πfτ dτ = ∑k =1
α k (t ) Ce τ rms
e − j 2πfτ k , (2.22)
−∞
In summary, in this thesis Jake’s channel model with an exponential power delay
profile is used to generate time- and frequency-variant channels as described in
(2.22).
2.3. SIMULATOR 33
It is usually good to have the autocorrelation matrix of the channel since most of the
more complex estimators use them. The autocorrelation matrix of the channel can
be obtained by the method presented in [9]. To derive the correlation function it is
assumed that α k (t ) has the same normalized correlation function rt (∆t ) for all k
and therefore it has the same normalized power spectrum. It follows that
{ }
rα (∆t ) ≡ E α k (t + ∆t )α k* (t ) = σ k2 rt (∆t ) ,
k
(2.23)
where σ k2 is the average energy of the k :th path and *

is the complex conjugate.
Using (2.22) and (2.23) it is possible to derive (2.24) the correlation function for the
frequency response for different frequencies and times, which is
{
rH (∆t , ∆f ) ≡ E H (t + ∆t , f + ∆f )H * (t , f ) }
 
= ∑ rα
k
k
(∆t )e − j 2π∆fτ k
= rt (∆t )


∑σ
k
2 − j 2π∆fτ k
ke
,


(2.24)
= σ H2 rt (∆t )r f (∆f )
where σ H2 is the total average power of the channel impulse response, which is
defined as
M −1
σ H2 ≡ ∑σ
k =0
2
k (2.25)
From (2.24) it follows that the correlation of expression (2.22) can be separated into
the correlation in time rt (∆t ) and the correlation in frequency r f (∆f ) . The
separation is possible since we adopted the channel model in (2.18). The correlation
in frequency is dependent of the multipath delay of the channel and is defined as
M −1
σ k2 − j 2π∆fτ k M −1 2 − j 2π∆fτ k
r f (∆f ) = ∑ k σ H2
e =
k
σke ∑ , (2.26)
As earlier mentioned the total power of the channel is assumed to be one, i.e.
σ H2 = 1 . The correlation in time rt (∆t ) is related to the relative speed of the
terminal, which is equivalent to the Doppler frequency and is defined as.
rt (∆t ) = J 0 (2πf d ⋅ ∆t ) , (2.27)
where J 0 is the zero-order Bessel function and f d is the Doppler frequency.

Using the expression for rt (∆t ) and r f (∆f ) it is possible to construct the
correlation matrixes for both time and frequency. For frequency the correlation
matrix is
r f [m] = r (m∆f ) , (2.28)
 r f [0] r f* [1] L r *f [N − 1]

 
 r [1] r f [0] L r *f [N − 2]
RHf Hf = f , (2.29)
M M O M
 
r f [N − 1] r f [N − 2] L r f [0] 
where m is each individual subcarrier and N is the total number of subcarriers.
The correlation in time can be obtained in the following manner,
rt [n] = J 0 (2πf d ⋅ T f n) , (2.30)
 rt [0] rt* [1] L rt* [T − 1]

 
r [1] rt [0] L rt* [T − 2]
R H t Ht = t , (2.31)
 M M O M 
 
rt [T − 1] rt [T − 2] L rt [0] 
where f d is the Doppler frequency, T f is the block length of one OFDM symbol,
J 0 is the zero-order Bessel function, n is each individual symbol and T is the total
number of symbols.
2.3.3 Signal to noise ratio
The signal to noise ratio is commonly defined as equation (2.32) in most systems
[8].
2
Ex h
SNR = , (2.32)
σ n2
where E x is the average energy of the transmitted bits, σ n2 is the variance of the
2
AWGN (Averaged White Gaussian Noise) and h is the energy of the channel.
The problem with this expression is for an OFDM system that the energy of the
channel is not constant and it varies significantly between the subcarriers. To
simplify the simulation an approximation was instead used that is based on the
average channel power of one OFDM symbol, as expression (2.33).
2.3. SIMULATOR 35
SNR avg =
[ ]
ExE h
2
(2.33)
σ n2
To simplify the simulations further the cyclic prefix effect on lowering the SNR is
neglected, since it will not give any further insight in the performs of the different
channel estimators.
2.3.4 Simulation settings
The simulation will only be conducted using the uplink frame with a bandwidth of
10 MHz as previously mentioned. The general settings for the simulator are
presented in Table 2.1.
Table 2.1: Information about uplink simulation setup

FFT Size 1024
Carrier Frequency 2.5 GHz
Null Sub-Carriers 184
Pilot Sub-Carriers 280
Data Sub-Carriers 560
Tiles 210
Symbol Period 91.4 µs
Frame Duration 5 ms
OFDM Symbols/Frame 48
Data OFDM Symbols 44
CP length 1/32 symbol
τ rms
0.3 µs
DL 24 symbols
TTG 1 symbols
UL 24 symbols
RTG 1 symbols
As seen in Table 2.1 the simulations are only carried out with a carrier frequency of
2.5 GHz, this is to limit the number of simulations. The WiMAX Forum is seeking
a permit for the 2.5 GHz in Sweden [14] and it is therefore the most interesting
frequency band. The cyclic prefix is chosen to be 1/32 OFDM symbols length, this
is motivated since the length of the multipath delay of the channel is never longer
than 1/32 OFDM symbols length.
The general layout of an uplink frame is presented in Figure 2.10. For a more
specific description of exactly which subcarrier is data- or a null-carrier, an
interested reader should read [1] and [2].
Figure 2.10: Layout of an uplink frame
The smallest structure in the uplink frame is a tile; which structure was presented in
the section 2.2.1. Figure 2.11 shows an illustration in how the tiles are positioned
together. In a WiMAX system the tiles are grouped together in groups of six [15].
These groups are formed by tiles across the entire spectrum, by a given permutation
scheme. These groups are referred to as slots. The key idea behind the creation of
the slots is to obtain frequency diversity. This aspect is not considered in this master
thesis, instead the frame allocation that is considered is described in section 4.1.
Figure 2.11: Tile positioning in frame
In IEEE 802.16e the pilot generator PRBS is initiated with a value that is based on
the frame number and also a few other parameters. This was considered too
complex to implement, instead the PRBS is initiated with “1010011101” for every
2.3. SIMULATOR 37
frame. Since the PRBS uses a fixed value for every new frame, all the pilot symbols
can therefore be precomputed and stored which speeds up the simulations.
Chapter 3
Channel estimation
This chapter describes the different channel estimation techniques that are analysed
in the thesis. It begins with the different channel estimation techniques and at the
end a few different interpolation methods are described.
Channel estimation in this master thesis is performed in three steps. The two first
steps are performed on the OFDM symbols containing pilot symbols and the last
step is preformed on the OFDM symbols that do not have pilot symbols. Recall
Figure 2.11, the vertical lines with pilot symbols is the OFDM symbols with pilot
symbols. The first step is to estimate the channel frequency coefficients at the pilot
symbols positions. Using the estimates of the channel frequency coefficients we
then interpolate over channel frequency coefficients corresponding to the data
symbols. Finally an interpolation in time over the different OFDM symbols is
required since not all OFDM symbols have pilot symbols.
3.1 1D estimator
An estimator that only uses information from one dimension; either time or
frequency is referred to as a 1D estimator. Three different 1D estimators in the
frequency dimension are investigated herein.
3.1.1 Least squares channel estimator
The LS (least squares) estimator is described in [3], the estimator finds the channel
impulse response ĥ LS , that minimize the squared error
2
ε = y − DWN hˆ LS , (3.1)
hˆ LS = arghˆ min(y − DWN hˆ LS ) H (y − DWN hˆ LS ) , (3.2)

LS
39
40 CHAPTER 3. CHANNEL ESTIMATION
where ε is the error between the received signal and the estimated signal, D has
the known symbols on its diagonal and has the dimension N × N , WN is the DFT
matrix which has the dimension N × N and y is the received vector and has the
dimension N × 1 . It follows that the estimated channel impulse response that
minimizes ε is
hˆ LS = QWNH D H y , (3.3)
where
Q = (WNH D H DWN ) −1 , (3.4)
Since we are interested in the frequency response of the channel at the subcarrier
frequencies, the expression can be rewritten by taking the DFT of the estimated
channel impulse response, according to.
LS = WN h LS = WN QWN D y
ˆ
H ˆ H H
(3.5)
By substituting (3.4) into (3.5), we obtain
−1
LS = D y
ˆ
H (3.6)
This estimation method is as mentioned above optimal in LS sense and in the case
of AWGN, it is also the ML estimate given one received OFDM symbol. In the
derivation it is assumed that all symbols that are transmitted are known to the
receiver on forehand. But this is not the case herein therefore D contains only the
pilot symbols and therefore has the dimension P × P , where P is the number of
pilot symbols. The data symbols are also removed form y prior estimation creating
y with dimension P × 1 .
In order to estimate all channel coefficients, the coefficients corresponding to the

data symbols position has to be interpolated. The interpolation method that is used
is described in section 3.4.1.
Since not all OFDM symbols have pilots, an interpolation in the time dimension is
also needed. The time interpolator is described in section 3.4.4. To illustrate this
better Figure 3.1 show how the estimator is implemented together with the
interpolators.
3.1. 1D ESTIMATOR 41
Figure 3.1: LS implementation
3.1.2 FIR interpolation algorithm
The approach for the FIR interpolation algorithm is that the channel impulse
response has a limited number of taps. It is therefore possible to estimate these taps
through the pilot symbols and then calculate the estimated frequency response of
the channel with the DFT. This method is presented in [3], but here some small
modifications are done is to suite the kind of uplink frame used in the simulations.
This method has it origin in the LS method described in section 3.1.1. Recall the
general expression of the channel frequency response of the LS estimator in (3.5).
LS = WN QWN D y ,
ˆ
H H H
(3.7)
in which Q is given by (3.4).
Assume that the channel impulse response h only has at most L taps. Let WP be
the matrix containing only the first L columns of WN . This will result in a partial
DFT matrix WP which has the dimension N × L . In order to make the calculations
faster, all rows in WP that do not correspond to a pilot symbol are set to zero. The
matrix containing the known pilot symbols D has the dimension N × N and has
the pilot symbols on its diagonal. All positions on the diagonal in D that those not
correspond to a pilot symbol is set to zero. The received vector y contains all
received symbols in one OFDM symbol and has therefore the dimension N × 1 .
With these modifications it is possible to rewrite the LS estimator as
Ĥ FIR freq = WP Q P WPH D H y


 −1
(3.8)
Q p = (WP D DWP )
H H
The inverse that is calculated to attain Q p is therefore performed on a matrix with

dimensions L × L . For this algorithm to work correctly, a few conditions have to be
met. First the number of taps in the channel impulse response L cannot be greater
than the number of pilot symbols. This is explained by the fact that at most L
unknowns can be solved using L equations. However if there are fewer taps in the
estimator than in the real channel there will be a model error. The number of taps
the FIR interpolation algorithm uses to estimate the channel has to selected or
estimated using some algorithm. In this master thesis the Akaike’s information
criterion [10] is used to determine the number of taps the FIR interpolation
algorithm estimates the channel with, the algorithm is more deeply described on the
next page.
Implementing the FIR interpolation algorithm in the frequency domain is

straightforward. It is simply done by taking the algorithm as described above,
OFDM symbol per OFDM symbol for the whole frame. But when implementing
the FIR interpolation algorithm it is important to keep in mind that not all OFDM
symbols have pilot symbols, therefore a time interpolator is required. The method
used to interpolate in time is described in section 3.4.4. To illustrate this better,
Figure 3.2 shows how the interpolator in time is implemented together with the FIR
interpolation algorithm.
Figure 3.2: FIR interpolation algorithm implementation
Estimation of number of taps

To be able to implement the FIR interpolation algorithm efficiently, the number of
required taps needs to be estimated or selected. This could be done in many ways,
herein the Akaike’s information criterion (AIC) described in [10] is used. The
Akaike’s information criterion is defined as followed
1 N 
AIC = N log
N ∑ε 2
(t, θ ) + γ ⋅ L , (3.9)
 t =1 
where N is the number of values that are compared, L is the number of taps the
estimator uses to estimate the true channel and thereby dim θ = L , γ is a
penalizing factor to penalize high order solution in the Akaike’s information
criterion γ = 2 and ε 2 (t , θ ) is the squared error between the received signal and the
transmitted signal given an estimator θ in the case of L FIR taps, ε (t , θ ) is
defined as
ε (t , θ ) = (d (t ) − dθ (t )) , (3.10)
where d (t ) is the sent symbol and dθ (t ) is the estimated symbol given an estimator
θ in the case of L FIR taps. Note, that this is another definition then (3.1), which
would correspond (3.11).
(t ,θ ) = y(t ) − yˆ (t ,θ ) = D(t )Wp (h(t ) − hˆ (t,θ )) + ~e (t ) (3.11)
The reason for using the soft decision in (3.10) is that it was simpler to implement.
The main advantage using the AIC is that the process of choosing the number of
taps for the filter can be automated. This way the specific number of taps that
minimizes AIC can be derived.
3.1.3 LMMSE
Another type of 1D estimator is the LMMSE (least minimum mean square error)
estimator, which is presented in [4]. Generally this estimator is described as
−1
 β  ˆ
LMMSE = R Hf Hf  R Hf H f +
ˆ
H I  H LS , (3.12)
 SNR 
where R H f H f is the correlation matrix of H , Ĥ LS is the least squares estimate

which is calculated as in section 3.1.1, SNR is the average SNR which is defined as
in (2.33) and β is modulation dependent and is defined as
{ }E 1d
2
2 
β = E dk , (3.13)
 k 
where d k is the symbol on the k:th subcarrier. For QPSK modulation, β = 1 .
There is one main problem with this algorithm, which is that it requires that all
symbols sent in an OFDM symbol are pilots. This can however easily be fixed by
combining it with the algorithm presented in [5]. The corresponding LMMSE
estimator is,
−1
 β  ˆ
LMMSE = R Hf Hpf  R Hpf Hpf +
ˆ
H I  H LS , (3.14)
 SNR 
where R Hf Hpf and R Hpf Hpf are the cross correlation and autocorrelation matrices
of the channel, which are defined as
[
R H f Hp f = E H f H pHf ] (3.15)
[
R Hpf Hpf = E H pf H pHf ] , (3.16)
where H pf is the channel response at the pilots position in the frequency domain
and H f is the channel response for the whole channel in the frequency domain. It
is possible to lower the complexity of the estimator by splitting up H f into two

different vectors, as below in expression (3.17). This idea is presented in [5].
Hence,
H p 
Hf =  f  , (3.17)
H pf 
where H pf is the channel response at the other locations in the frequency domain
than the pilots. With this approach it is possible to separated the estimation and
interpolation into two different expressions as follows,
−1
 β  ˆ
p f , LMMSE = R Hp f Hp f  R Hp f Hp f +
ˆ
H I  H LS (3.18)
 SNR 
−1
pf ,LMMSE = R H pf Hpf R Hpf Hpf H pf ,LMMSE
ˆ
H ˆ (3.19)
The interpolator is more deeply described in section 3.4.3, the focus in this section
is on the estimator, expression (3.18). A potential problem with the estimator is the
case of high SNR, since the inverse in the expression (3.18) will converge to
R −Hp
1
f Hpf
. This can be a problem since R Hpf Hpf generally does not have full rank.
Since the impulse response of the channel typically is of finite duration with most
of its energy limited to a few of its taps. However a solution to this is presented in
[5]. It is based on the singular value decomposition of
R Hp Hp = UVU H , (3.20)
f f
where V is a diagonal matrix with the eigenvalues on its diagonal and U is unitary
matrix with the corresponding eigenvectors. The matrix V is transformed so that it
only contains k non-zero eigenvalues and U is also transformed so that it only
contains the corresponding eigenvectors thereby forming a k-point DFT-matrix
WK [5]. With these modifications the LMMSE estimator can be written as
δ1 0
 WHH
pf ,LMMSE =
ˆ
H W O ˆ (3.21)
K  K ls
 0 δ k 
where
 λi
, λi ≠ 0
 β
δ i =  λi + (3.22)
 SNR
0, λi = 0
The LMMSE estimator time interpolator is described in section 3.4.4. Figure 3.3
illustrates how the LMMSE estimator is implemented together with the
interpolators.
Figure 3.3: LMMSE implementation
3.2 2D estimator
So far, the presented estimators only utilize the correlation between the subcarriers
within a time instance for estimation. It is possible to get a better estimation in
MMSE sense if the channel estimator uses information from both the frequency
dimension and the time dimension, at which point it becomes a 2D estimator.
3.2.1 2D MMSE channel estimator
The 2D MMSE channel estimator is described in [5]. The estimator estimates the
channel coefficients frame by frame. The 2D MMSE channel estimator uses a LS
estimator that estimates an entire frame, according to
Hˆ LS (m,n) = Y(m,n) D(m,n) , (3.23)
where Hˆ LS (m,n) is the LS estimate tensor for one frame, Y(m,n) is a tensor
containing the received symbols for one frame, D(m,n) is a tensor containing the
sent symbols for one frame and m together with n defines the subset of all the pilot
symbols in the uplink frame. The 2D MMSE channel estimator is as given in [5]
Hˆ 2DMMSE(l,i) = ∑ w(l,i,m,n)Hˆ
(m,n)∈P
LS (m,n) , (3.24)
where Hˆ 2DMMSE(l,i) is the 2D estimate tensor of the channel frequency response for
the entire frame and w(l,i,m,n) is a 4D tensor which is obtained by minimizing
 2
w(l,i,m,n) = arg w minE  h(l,i) − Hˆ 2DMMSE(l,i)  (3.25)
 
Using the orthogonality principle [5] the channel estimator can be derived by using
that.
[( )
E h(l,i) − Hˆ 2DMMSE (l,i) Hˆ *LS (m,n) = 0 ] (3.26)
Combining it with the expression (3.24) yields expression (3.27)
[
E h(l,i)Hˆ *LS (m,n) = ]
∑ w(l,i,l ′,i′)E [Hˆ ]
(3.27)
LS (l ′,i ′)H LS (m,n)
ˆ*
l ′,i ′∈P
This is a system of linear equations and it is now possible to calculate the

coefficients w(l,i, l ′,i ′) that minimize equation (3.25). However this requires that the
[ ] [ ]
cross-correlation E h(l,i)Hˆ *LS (m,n) and the auto-correlation E Hˆ LS (l ′,i ′)Hˆ *LS (m,n)
are known. This filter was considered to complex to implement given the size of the
cross-correlation, auto-correlation and the coefficient matrices. Instead a 2D
estimator with lower complexity was implemented, which is described in section
3.2.2.
3.2.2 Low complexity 2D channel estimator
Our approach to construct a low complexity 2D estimator is to use two different 1D

estimators; i.e. one that estimate the channel in the time dimension and one that
estimate the channel in the frequency dimension. Such an estimator is presented in
[5], where two LMMSE channel estimators are used, first one in the time dimension
and then one in the frequency dimension. The LMMSE estimator is the same as the
one already described under the chapter 1D estimator, in section 3.1.3. The time
dimension LMMSE estimator can be derived in the same way as the LMMSE
frequency dimension channel estimator already described in section 3.1.3.
Therefore, the low complexity 2D estimator time estimator is
−1
 β  ˆ
t, LMMSE = R H t Hp t  R Hp t Hp t +
ˆ
H I  H LSt , (3.28)
 SNR 
where Ĥ LSt is the LS estimate in the time dimension, R Hp t Hp t and R HtHpt is the
autocorrelation and cross correlation of the channel, defined as
[
R Hp t Hp t = E H p t H pHt ] , (3.29)
[
R HtHpt = E H t H pHt ] , (3.30)
where H t is the channel frequency response in the time dimension and H p t is the
channels frequency response in the time dimension at the pilot symbols positions.
The low complexity 2D estimator frequency estimator is
δ1 0
ˆ
H = W  O W H H
ˆ (3.31)
pf ,LMMSE K  K LS
 0 δ k 
where
 λi
, λi ≠ 0
 β
δ i =  λi + (3.32)
 SNR
0, λi = 0
−1 ˆ
pf ,LMMSE = R Hpf Hpf R Hpf H pf ,LMMSE
ˆ
H (3.33)
Expression (3.31) and (3.33) has the same form as the 1D LMMSE estimators
(3.18) and (3.19), considered in section 3.1.3. Figure 3.4 illustrates how the two
LMMSE estimators are implemented on the uplink frame in the simulations.
3.3. ADAPTIVE ESTIMATOR 49
Figure 3.4: Low complexity 2D implementation
3.3 Adaptive estimator

An adaptive estimator is an estimator that updates its parameters over time as a
consequence of changing channel statistics. This is a fundamental difference
compared to the 1D and 2D estimator that has been presented earlier which are
static in that sense. In this section a low-complexity Kalman estimator will be
described, but the section starts with a closer look at the ordinary Kalman estimator.
3.3.1 Kalman estimator
As an approach to estimate the channel, a first order autoregressive process is

assumed, the noise is Gaussian wide-sense stationary uncorrelated scattering
(GWSSUS) and the channel has uniformly distributed angle of arrival. A channel
with uniformly distributed angle of arrival is what is used for the simulation, see
section 2.3.2. The AR-model that is assumed to model the time-variant channel is
H (l + 1) = F H (l ) + v (l ) , (3.34)
where H (l ) is the frequency response of the channel at the pilot symbols positions
with dimension P ×1 and v(l ) is the driving noise at the pilot symbols position
with dimension P ×1 , which is a zero-mean process that satisfies
[ ] V, m = 0
E v (l ) v H (l + m ) =  . (3.35)
0, m ≠ 0
Using this AR-model it is possible to construct the following Kalman channel

estimator that is optimal in an LMMSE sense, as defined in [7].
[
K (l ) = FP (l , l − 1)C H (l ) C(l )P (l , l − 1)C H (l ) + σ 2 I ]
−1
(3.36)
(l ) = Y(l ) − C(l )H
ˆ (l ) (3.37)
ˆ (l + 1) = F H
H ˆ (l ) + K (l ) (l ) (3.38)
[ ]
P(l + 1,l ) = F I − F −1K (l )C(l ) P(l,l − 1)F H + V , (3.39)
where C(l ) is a P× P diagonal matrix with pilot symbols on its diagonal where P
is the number pilot symbols and σ 2 is the variance of the AWGN. To estimate the
whole frequency channel response this method require some interpolation method
3.3.2 The low-complexity Kalman estimator
The Kalman estimator described above is in most cases impossible or impractical to

implement. Therefore a low-complexity Kalman estimator was implemented, which
is described in [7]. The method assumes that a circular constellation is used; this is
true in this case since BPSK is used for the pilot symbols. As approach we model
the correlation of the time-variant channel as
[ ]
 P J (2πf mT ), i = j
E h(l , i )h∗ (l + 1, j ) =  i 0
,i ≠ j
(3.40)
0
[
where Pi = E h(l , i )
2
], f m is the maximum Doppler shift, T is OFDM symbol
time and J 0 (•) is the Bessel function of the first-kind and zero order. Using that
H (l ) = Wh(l ) , (3.41)
we obtain
[ ]
E H (l )H H (l + 1) = J 0 (2πf mT ) WPW H (3.42)
where P is a N × N diagonal matrix with diagonal elements Pi and W is a M× N

partial DFT matrix obtained from a DFT matrix by deleting the rows that does not
correspond to pilot symbols. Here M denotes the number of pilot symbols in one
OFDM symbol and N is the total number of symbols in one OFDM symbol. In the
AR-model in (3.34), F is modelled as
3.3. ADAPTIVE ESTIMATOR 51
F = J 0 (2πf mbT )WPW H , (3.43)
where b is a design parameter that determines the memory of the algorithm and is
more deeply described in [7]. As earlier mentioned the driving noise v(l ) in the
AR-model is a zero-mean process that satisfies
[ ]
V, m = 0
E v (l ) v H (l + m ) =  . (3.44)
0, m ≠ 0
Here the definition of V is as in [7].
V = (1 − J 0 (2πf mbT ) 2 )WPW H (3.45)
Using these definitions together with the AR-model it is possible to construct the
low-complexity Kalman estimator. The main idea behind the low-complexity
Kalman estimator is to factorise P (l , l − 1) using eigenvalue decomposition, which
gives
P(l,l −1) = UD(l )U H , (3.46)
where U is the unitary matrix whose columns is the eigenvectors and D(l ) is a
diagonal matrix with the eigenvalues on its diagonal. An induction proof that the
unitary matrix U is the same for any l is presented in [7]. Based on this proof it is
possible to construct the following Kalman estimator.
(l ) = Y(l ) − C(l )H
ˆ (l ) (3.47)
ˆ (l + 1) = J (2πf KT ) ⋅
H 0 m
[ ]
(3.48)
Hˆ (l ) + UD(l )(D(l ) + σ 2 I ) −1 U H C H (l ) (l )
M
To lower the complexity it is possible to update only the diagonal in D , i.e. the
eigenvalues of P (l , l − 1) instead of updating the whole diagonal matrix D .
J 0 (2πf m KT ) 2 σ 2 λi (l )
λi (l + 1) = +
λ i (l ) + σ 2 (3.49)
(1 − J 0 (2πf m KT ) )λi (1)
2
Since the channel model used in this master thesis has a finite impulse response it is
possible to reduce the number of eigenvalues in D to the ones that are nonzero and
also taking into consideration that not all subcarriers carry pilot symbols U can be
reduced to an MxJ matrix where J is the number of nonzero eigenvalues in D .
To study how the interpolation algorithm in the frequency dimension affects the
symbol error two different frequency interpolation methods were implemented
together with the Kalman estimator. To distinguish between the two interpolations
algorithm the two implementations of the low-complexity Kalman estimator are
referred to as,
• Low-complexity Kalman method 1, interpolation method in section 3.4.1
• Low-complexity Kalman method 2, interpolation method in section 3.4.2.
The time interpolation algorithm described in section 3.4.4 is used with the both
implementations. Figure 3.5 illustrates how the frequency interpolation and time
interpolation methods are implemented together with the Kalman channel
estimators.
Figure 3.5: Low-complexity Kalman implementation
3.4 Interpolation
To estimate the channel frequency response for the entire frame in this thesis
require one interpolator over each individual OFDM symbol in the frame i.e. in the
frequency dimension and one interpolator between different OFDM symbols i.e. in
the time dimension of the frame. There are three different interpolators considered
in the frequency dimension and one interpolator in the time dimension.
3.4. INTERPOLATION 53
3.4.1 Interpolation method 1 in frequency
Interpolation method one in frequency is a Wiener filter that is proposed in [8],

which is
−1
interpolat edf = ( R Hp f Hp f R Hp f H f ) H P ,
ˆ
H H ˆ
(3.50)
where Ĥ interpolat edf is the interpolated channels frequency response, Ĥ p is the

estimated channels frequency response at the pilot symbols position, R Hp f Hp f and
R Hp f H f is derived from the correlation matrix in the frequency dimension R H f H f
as follows.
 rH p1 H p1 rH p1 H p 2 K rH p1 H pN 
 
 rH H rH p 2 H p 2 L rH p 2 H pN 
R Hp f Hp f =  p 2 p1 , (3.51)
M M O M
 
 rH H rH pN H p 2 L rH pN H pN 
 pN p1 
where H pX denotes which pilot position in the OFDM symbol and N is the total
number of pilots.
 rH p1 H1 rH p1 H 2 K rH p1 H M 
 
 rH H rH p 2 H 2 L rH p 2 H M 
=  p2 1
M 
R Hp f H f , (3.52)
M M O
 
 rH H rH pN H 2 L rH pN H M 
 pN 1 
where H pX denotes which pilot position in the OFDM symbol, HX denotes which
subcarrier and M the total number of subcarriers. To obtain the correlation matrix
R H f H f the method proposed in [9] is used, which is described in section 2.3.2.
The interpolator has the same potential problem as the LMMSE estimator with the
inverse of R Hpf Hpf , since R Hpf Hpf generally does not have full rank. This is
solved using a so-called pseudo-inverse [10]. The first step is to do a singular value
decomposition (SVD) of R Hpf Hpf , i.e.
R Hp f Hp f = UVU H (3.53)
Above, U is a unitary matrix with the eigenvectors in its columns and V is

diagonal matrix containing the eigenvalues. The pseudo-inverse is calculated by
remove the non-zero eigenvalues from V so it only contains k non-zero
eigenvalues and also reducing U to contain only k eigenvectors and thus creating
the matrix U K . Hence the pseudo-inverse of R Hpf Hpf is
−1
λ1 0
−1  
R Hp f Hp f
= UK  O H
 UK , λi ≠ 0 (3.54)
 0 λK 
The second interpolation method that was implemented is proposed in [11] and is
based on filtering the frequency channel response with a low-pass filter. The
interpolation method is implemented in Matlab®, as the function interp. The
Matlab® implementation assumes that the spectrum of the interpolated signal has a
flat spectrum. In the simulations of the estimator the Matlab® implementation is
used, although the frequency response of the estimated channel does not have a flat
spectrum. This interpolation method can still be used for comparison reasons with
the other interpolation methods.
The interpolation begins with inserting N int − N P evenly spaced zeros into Ĥ p
creating Ĥ zero , where N P is the number of given pilots and N int is the total
number of carriers. A requirement on N int is that N int = r , where r should be

NP
an integer. Then a FIR filter is created that when applied on Ĥ zero lets the original
values pass through unchanged and interpolates the other values so that the mean
square error is minimized, based on the assumption the spectrum is flat. Each
interpolated value is based on 2 L original values. This creates a FIR filter that has
the length 2rL + 1 taps. The simulations carried out in this thesis used r = 2 and
L=4.
The FIR filter g FIR (n) , is created by splitting up g FIR (n) into r sequences
g FIRτ (l ) = g FIR (rl + τ ), τ = 0,1,..., r − 1 and then solving the following equations for
every τ
L −1
∑g
l =−L
FIR (rl + τ )Φ(r (l − m )) = Φ(rm + τ )
(3.55)
m = − L,..., L − 1
g FIR = , (3.56)
where is the autocorrelation of the vector that is to be interpolated. In our case

this would be Ĥ p , but the Matlab implementation of the algorithm assumes that
3.4. INTERPOLATION 55
Ĥ p has a flat spectrum. Finally, the vector Ĥ zero is filtered with the FIR filter
g FIR (n) resulting in the vector Ĥ int .
Interpolation method 3 in frequency is very closely related to interpolation

method 1 in frequency. However the difference is that interpolation method 3 in
frequency only interpolates the non-pilot symbols compared with interpolation
method 1 in frequency that interpolates the whole frequency response of the
channel. Interpolation method 3 in frequency is derived from the LMMSE
estimator, in section 3.1.3. It is defined as
ˆ =R
H p Hp R −Hp
1
f Hp f
ˆ ,
H p (3.57)
f Hp f
where R Hp f Hp f is defined as expression (3.51), Ĥ p f is the interpolated channels

frequency response at the non-pilot symbols position, Ĥ p f is the estimated
channels frequency response at the pilot symbols position and R H pf Hp f is the cross
correlation between the non-pilot position and the pilots position which is derived
from the general correlation matrix in the frequency dimension R H f H f as follows
 rH p1 H p1 rH p1 H p 2 K rH p1 H pN 
 
 rH H rH p 2 H p 2 L rH p 2 H pN 
R Hp =  p 2 p1 , (3.58)
f Hp f M M O M
 
 rH H rH pN H p 2 L rH pN H pN 
 pN p1 
where H pX denotes which pilot position in the OFDM symbol, H pX denotes which
non-pilot position in the OFDM symbol and N is the total number of pilots.
This interpolation method has the same potential problem in frequency as

interpolation method 1. That is the inverse of R Hp Hp , since R Hp Hp generally
f f f f
does not have full rank. The solution to this problem is presented in section 3.4.1. In
short the solution is based on calculating the single value decomposition of
R Hp Hp and then calculating the pseudo-inverse [10]. This results in the following
f f
expression
−1
λ1 0
−1  
R Hp f Hp f
= UK  O H
 UK , λi ≠ 0 , (3.59)
 0 λK 
where λ K is the K:th non-zero eigenvalues in R −Hp

1
f Hp f
, and U K is a unitary
matrix containing the corresponding eigenvectors.
3.4.4 Interpolation method 1 in time
Since not every OFDM symbol have pilot symbols, a simple interpolator in the time
dimension was designed. The interpolator is based on the assumption that OFDM
symbols that are close in time have a strong correlation of there frequency channel
response. The interpolator in time is based on interpolating a whole frame. The
interpolation begins with creating a matrix with the same size as the entire
uplink frame and zeros on all positions. Then all the estimated and interpolated
frequency response vectors Ĥ for that frame are inserted on their corresponding
positions in . The following filter is then applied per element in the matrix for
every i and n
(i, n − 1) + (i, n + 1)
(i , n) = , (3.60)
2
where n is the index of all the OFDM symbols without pilots and i denotes each
subcarrier.
3.5 Summary
Below is a short summary of the different algorithms studied together with a
description how they are linked together.
3.5.1 1D estimators
The first 1D estimator to be analysed is a LS estimator. The LS estimator is

particularly interesting since it is one of the most simple estimation algorithms that
exist. The estimation algorithm is than linked with interpolation method 1 in
frequency and interpolation method 1 in time. This achieves an interesting
combination since the estimation algorithm is very simple and the interpolation
method 1 in frequency is more complex as it uses correlation properties to
interpolate in frequency.
The second 1D estimator, the FIR interpolation algorithms is of interest since it has
another approach to the estimation and interpolation in frequency then the other
estimators and interpolators studied. Since it estimates the taps of the channel in the
time dimension first and from this it calculates the frequency response of the
channel. To be able to estimate the whole channel frequency response for the frame
this algorithm was linked together with interpolation method 1 in time.
The third 1D estimator is the LMMSE estimator and this is of interest since it has a
more complex estimation algorithm that uses correlation properties of the channel.
This algorithm is linked together with interpolation method 3 in frequency which
3.5. SUMMARY 57
also uses correlation properties to interpolate. Finally it uses interpolation method 1

in time to interpolate.
3.5.2 2D estimation
The only 2D estimator that is studied is the low complexity 2D estimator. It is

interesting to study a 2D estimator to compare its result with the different 1D
estimators and the adaptive estimators.
3.5.3 Adaptive estimation
The adaptive estimator that is studies is a low complexity Kalman estimator. A

Kalman estimator is a common adaptive estimator and it is interesting to see if an
adaptive estimator can perform better than the other estimators.
To study briefly how the choice of interpolator in frequency affect the result two
different frequency interpolators in frequency is implemented together with the low
complexity Kalman estimator. The two different interpolators are frequency
interpolation method 1 and frequency interpolation method 2. Since we want to
study the affect the interpolator in frequency has, both implementations used
interpolation method 1 in time.
Chapter 4
Simulations
4.1 The Test cases

When simulating a system there are often many parameter configurations that can
be interesting to investigate, but it is often necessary to reduce the number of test
cases to save time. The following three test cases were simulated.
• Case 1 a whole uplink frame, section 4.1.1

• Case 2 a narrow rectangle in time, section 4.1.2
• Case 3 a narrow rectangle in frequency, section 4.1.3
The reason for choosing these three test cases are that in an OFDMA system, a
terminal is given a number of subcarriers during a certain time interval to
communicate with the base station. The three test cases represent three basic types
of frame allocations. It is interesting to investigate if the channel estimators perform
differently on different frame allocations. Of course, other types of frame
allocations are possible, but to keep the number of simulations on a reasonable
level, these three test cases are the only ones that have been considered.
All the three test cases were simulated using QPSK, 16QAM and 64QAM, which
are the three different types of modulations supported by IEEE 802.16e. The SNR
is within 1-35 dB. However the result is only presented for the 64 QAM simulations
since the other modulations did not give any more information in respect to the
performance of the estimators. The simulations were also carried at 0 km/h,
30 km/h and 120 km/h. All the simulations were carried out over 500 frames to get
a good average result. The goal of the simulations was to determine which type of
estimator that shows the best performance for the different frame allocations and
relative speed. Although some of the simulation settings are very unlikely in a real
system, the goal is also to test how the estimators behave during extreme
conditions.
The channel model used in the simulations is described in section 2.3.2. The
channel has 10 different delay paths and has the average total power gain of one so
59
60 CHAPTER 4. SIMULATIONS
that it does not change the average power of the signal. A summary of the settings
for the simulations is given in Table 4.1.
Table 4.1: Settings for the simulations

Speeds 0 km/h No speed 30 km/ Mid speed 120 km/h High speed
Channel taps 10 taps
Modulation 64 QAM
SNR 1-35 dB
Simulations 500 frames
4.1.1 Specification case 1
Table 4.2: Settings for case 1

Subcarriers 93-933
Time 1-24
Data symbols per frame 13440
Pilot symbols per frame 6752
Figure 4.1: Simulation case 1

Subcarriers 401-500
Time 1-24
4.1. THE TEST CASES 61

Subcarriers 93-893
Time 1-3

4.2 Tuning of channel estimators

The first thing to do before comparing the different estimators is to tune the
different channel estimators. Every channel estimator except the LS estimator does
have at least one tuneable design parameter.
In the case of the FIR interpolation algorithm, the number of taps the estimator
should estimate must be selected or estimated. To determine the required number of
taps the Akaike’s information criterion [10] is used with a penalizing factor of 4.
For more information on Akaike’s information criterion see section 3.1.2.
The LMMSE 1D section 3.1.3, and the low complexity LMMSE 2D section 3.2.2,
estimator does have the SNR value as a design parameter. For these estimators the
true SNR value is used. When implementing any of these estimators in a real
system an SNR estimator has to be implemented.
The low-complexity Kalman estimators and low-complexity LMMSE 2D requires

the Doppler frequency and the variance of the noise. Instead of estimating the
Doppler frequency and the variance of the noise, as would be done in reality, the
estimators use the true values.
The low-complexity Kalman estimators have the design parameter b, which is

linked to the memory of the estimator. To determine b simulations have been
carried out for each test case. The same conclusion is possible to draw from all the
three different test cases, therefore only case 1 is presented. The simulations were
carried out according to Table 4.1, with the exception that they were only simulated
for 10 dB and 25 dB in SNR.
4.2. TUNING OF CHANNEL ESTIMATORS 63
Figure 4.4: High speed simulation parameter b
Figure 4.5: Mid speed simulation parameter b

Figure 4.6: No speed simulation parameter b
As seen by Figure 4.4, Figure 4.5 and Figure 4.6 the optimal b is clearly speed
dependent and it seems not to change with respect to SNR. To simplify the
estimations the same b will be used for all three different speeds in the three test
cases. The optimal b is the one with the lowest SNR value. The decision was made
to use b=2, because some initial simulation had already been carried out with b=2
when the parameter b was simulated and since b=2 is relative close to the optimal
value of b.
An explanation to why Figure 4.6 has another shape than Figure 4.4 and Figure 4.5
is that the parameter b is cancelled out by a zero Doppler frequency in the
eigenvalue update equation. See section 3.3.2 for details.
4.3 Case 1
4.3.1 Simulation case 1
With the tuning setting for each filter as stated in section 4.2, the simulations were
carried out according to the Table 4.1 and Table 4.2.
4.3. CASE 1 65
Figure 4.7: Case 1 no speed simulation part 1

Figure 4.9: Case 1 mid speed simulation part 1

4.3. CASE 1 67
Figure 4.11: Case 1 high speed simulation part 1

4.3.2 Conclusion case 1
As seen by the results in Figure 4.7 to Figure 4.12, the FIR interpolation algorithm
shows the best performance followed by the low-complexity Kalman with method
2.
Further more it can be noticed that the two LMMSE estimators perform about the
same. This could be explained by the fact that there is little information to gain by
using the time dimension, since there are only 16 pilots in the time dimension
compared with 420 pilots in the frequency dimension for a full uplink frame.
Therefore a very simple time interpolator as the case of the LMMSE 1D estimator
gives the same result as a complex time estimator as in the case of the low-
complexity LMMSE 2D estimator.
All channel estimators loss performance compared with the true channel as the SNR
grows except the FIR interpolation algorithm. What all the other estimators have in
common is that they all have at least one method that is based on the knowledge of
the averaged energy per tap. The averaged energy per tap is used to create the
correlation matrices that are used in the LS estimator, LMMSE 1D estimator, low-
complexity LMMSE 2D estimator and low-complexity Kalman method 1. The
averaged energy per tap is also used in both the Kalman estimators as the
underlying knowledge for some of the matrices. A theory is that since not the
specific energy per tap for each OFDM symbol is used there will be small
estimation errors compared with the FIR interpolation algorithm that only uses the
information about current frequency response to estimate the channel frequency
response.
An interesting observation is the fairly large difference in performance between the

two interpolation algorithms with the low-complexity Kalman estimator. In this
simulation case interpolation method 2 outperforms interpolation method 1, which
can especially be seen in Figure 4.8, but also in Figure 4.10 and Figure 4.12.
4.4 Case 2
4.4. CASE 2 69


4.4. CASE 2 71

4.4.2 Conclusions case 2
The result in case 2 is different from case 1. Because in case 2 (Figure 4.13 to
Figure 4.18) the FIR interpolation algorithm performs about the same as the two
LMMSE algorithms and low-complexity Kalman method 1.
By comparing the figures from case 2 (Figure 4.13 to Figure 4.18) with
corresponding figures for case 1 (Figure 4.7 to Figure 4.12), it can be concluded
that the LMMSE 1D, LMMSE 2D, Least squares and the low-complexity Kalman
method 1 performs better in case 2 than in case 1. This is an interesting fact that
seems to be linked to the interpolation method used for these estimators, which is
what the different methods, has in common. The difference between case 1 and
case 2 is the number of pilot symbols and also the number of interpolation points. A
theory to why we get this result is that there is a small error in the correlation
matrixes used for interpolation, especially in how a certain interpolated symbol is
linked to an estimated symbol far away. This would explain the relative gain in
performance for the above mentioned estimators, since it is relatively narrow in
frequency compared to case 1. The theory is that the error in the correlation
matrices arises because of the averaged power per tap is used to generate it and
therefore it will only be optimal in an averaged sense and not in a specific case.
A possible implementation for estimating and interpolating a whole frame could be

to split the frame into smaller sub frames, where each sub frames is estimate and
interpolated individually. This would take advantage of the fact the above
mentioned estimators perform better on fewer subcarriers. The proposed method
would also lower the total number of computation of the estimators.
Another conclusion that is possible to draw is that the FIR interpolation algorithm
performs slightly worse in case 2 than in case 1. This is because there are fewer
pilot symbols to use for estimation, hence a worse performance.
The low-complexity Kalman method 2 is the worst estimator in case 2 which was
not the case in case 1. One theory to why this occurs is two folded. For the first, all
the estimators that use correlation properties to interpolate perform better, which is
also the case for the low-complexity Kalman method 1. Secondly, the interpolation
method in frequency for the low-complexity Kalman method 2 degrades due to the
frame allocation in case 2.
4.5 Case 3
4.5. CASE 3 73


4.5. CASE 3 75

4.5.2 Conclusion case 3
The results from case 3 are almost the same as for case 1, where a whole frame was
used. This follows by comparing Figure 4.19 to Figure 4.24 with Figure 4.7 to
Figure 4.12. Hence, the same conclusion holds for this case as well, namely that the
FIR interpolation algorithm performs best and is followed by the low-complexity
Kalman with method 2 used as interpolator, which performs almost identical as in
case 1.
Since neither of the two the low-complexity Kalman estimators degrades in

performance. It can be concluded that the convergence of the low-complexity
Kalman estimator is rather fast. This is linked to the memory parameter b, see
section 4.2.
4.6 General Conclusions

An effect that can be observed in all the simulations is that the two LMMSE
algorithms converge to the same value as the LS algorithm in the high SNR region.
This is due to the fact that at high SNR values expression (3.18) converges to
expression (4.1).
−1
 β  ˆ
p f , LMMSE = R Hp f  R Hp f +
ˆ
H I  H LS
 SNR  (4.1)
SNR
 → R Hp f R −Hp
→∞
1 ˆ
f
H LS = H
ˆ
LS
−1 ˆ
pf ,LMMSE = R H pf Hpf R Hpf H pf ,LMMSE
ˆ
H (4.2)
Although the interpolation is not identical in the LMMSE algorithm as for the LS
algorithm, it is still based on the use of correlation properties of the channel.
Therefore the algorithm can be considered equal at high SNR.
The different channel estimators depend on different amount of knowledge about

the channel. In Table 4.5 the different channel estimators are presented together
with that knowledge they require about the channel.
4.6. GENERAL CONCLUSIONS 77
Table 4.5: Properties that channel estimators utilize

Number Correlation
Average Correlation
of taps Doppler of channel
energy of channel SNR
in frequency in
per tap in time
channel frequency
LS S X
FIR X
LMMSE
S X X
1D
LMMSE
S S X X X
2D
Kalman
X X X X
method 1
Kalman
X X X
method 2
* Required design parameter (X) and (S) marks parameter that is required due to this particular
implementation
The two methods that does only depend on one property is the LS and FIR
interpolation algorithm, although the LS method has a sub depends due to the
interpolation method.
A general conclusion that is possible to draw is that there is no gain in performance

by using the low-complexity LMMSE 2D estimator compared to the LMMSE 1D
estimator. An explanation to this is that are quite few pilots in the time dimension
compared to the frequency dimension combined with that variations in frequency
dimension is larger than the variation in the time dimension over one frame. Hence
the time interpolator described in section 3.4.4 is sufficient.
The estimator that generally performs best is the FIR interpolation algorithm. In
case 2 there is a slight degradation in performance compared to the other estimators.
Since the FIR interpolation algorithm calculates the energy for each tap, based on
one OFDM symbol and in case 2 there is fewer pilot symbols to estimate the
channel with.
One of the most interesting results is that interpolation method 2 in frequency

performs better than interpolation method 1 in frequency in case 1 and case 3, but
not in case 2. This is possible to analyse when comparing the low-complexity
Kalman method 1 with the low-complexity Kalman method 2. Although a
theoretical proof to why this occurs is not reached, it is still possible to reason why.
Since the only thing that changes between the cases are the number of pilots and
interpolation points, it is therefore linked to it. The theory is that there is a small
error in the correlations matrixes for how an interpolation point is dependant on a
pilot symbols far away. Since in case 2 the allocation in the frame is more compact
in frequency than in case 1 and case 3, this error will not affect the interpolation
that much compared to in case 1 and case 3. This small error arises because the
correlation matrixes are based on the averaged tap power and is therefore only
optimal in an averaged sense and not in every individual case.
The good result in the simulation in case 2 for some of the estimators raises the idea
that it would be possible to estimate and interpolate a whole frame by splitting it up
in sub-areas and than estimating and interpolate each sub-area separately. This
would also lower the total number of computations needed to estimate a whole
frame.
Another conclusion that can be made is that the choice of interpolation method is
very important. For example when comparing the low-complexity Kalman method
1 with the low-complexity Kalman method 2, is that in almost all the simulation the
two low-complexity Kalman methods begin to differentiate at around 20 dB SNR
and at 35 dB the difference is relatively significantly.
All the estimators perform about the same for SNR lower than 15 dB. This is an
interesting property which means that the choice of channel estimator is not that
important in terms of symbol errors for low SNR. When choosing a channel
estimating method for low SNR the focus should instead be on how much
information the estimating methods needs and also how high its complexity is.
To get a basic idea what the complexity difference is between the different
methods. The Matlab® command profile was used to measure the time to execute
each algorithm. This does not give an absolute truth in terms of complexity,
because it is linked to how the estimators are implemented and to some extent
platform depend, but it still gives a basic idea of the complexity.
Table 4.6: Case 1 profile

Estimation method Percent compared
to slowest
Kalman method 1 100 %
FIR 61 %
LMMSE 2D 24 %
LMMSE 1D 15 %
LS 9%

to slowest
FIR 100 %
Kalman method 1 5.7 %
Kalman method 2 5.4 %
LMMSE 2D 3.2 %
LMMSE 1D 2.3 %
LS 2.0 %
4.6. GENERAL CONCLUSIONS 79

to slowest
FIR 70 %
LMMSE 2D 28 %
LMMSE 1D 17 %
LS 10 %
The FIR interpolation algorithm that generally performs the best in the simulation
and does only depend on one property has a relatively high complexity compared
with the other methods, especially in case 2, Table 4.8, where it performs far worse
than all the other estimators. This is because the matrices in the FIR algorithm do
not change in size when the frame allocation is changed. However the matrices of
the other algorithms change in size and therefore the number of computation
change, hence they perform relatively much faster in case 2.
A noticeable difference in computation time is that between the two LMMSE

estimators, where LMMSE 2D is far slower than the LMMSE 1D. This is an
expected result since the LMMSE 2D estimates also in the time dimension.
However when combining this with the result of the simulations it is possible to
conclude that this extra computation time does not give a better estimate.
Chapter 5
Conclusions and Future work
This master thesis has focused on different channel estimation techniques in an

OFDM system with parameters from IEEE 802.16e. Several general parameters has
been discussed and analysed to give an insight to the channel estimation.
To analyse the different channel estimation techniques three cases with different
frame allocations was defined. The three allocations were a whole frame, a narrow
rectangle in time and a narrow rectangle in frequency. The simulations were also
conducted at three different relative velocities.
The channel estimators in the OFDM system considered in this thesis consists of
three steps. First the channel is estimated at the pilot symbols location, second the
channel is interpolated over the pilot symbols in the frequency dimension and last it
requires to be interpolated in the time dimension. The order of the two last steps
may change depending on the estimation method.
The frame allocation had a greater impact on the relative performance between the
different channel estimators than the relative velocity. Even if the channel
estimator’s performance degraded with higher relative velocity, their mutual order
in terms of symbol error did not change; this was the case if the frame allocation
was changed.
In general the channel estimator that performed the best in terms of lowest symbol
error is the FIR interpolation algorithm. This method tries to estimate the energy of
the channel taps. However this algorithm as it is presented has a rather high
complexity, especially when it estimates a sub-area in a frame that as few
subcarriers as case 2.
One of the most interesting properties that were discovered is the big impact the
interpolation method has over the estimating method. Two different channel
estimators based on the low-complexity Kalman method, but with different
interpolation methods were analysed. The performance between the two methods
was significantly different for SNR above 25 dB.
81
82 CHAPTER 5. CONCLUSIONS AND FUTURE WORK
The fact that many of the channel estimators performed relatively well in case 2 it
raises the idea that an entire frame can be estimated and interpolated by splitting it
into small areas that are estimated and interpolated separately. This would also
lower the computation cost to estimate the whole frame.
The amount of configuration parameters that the channel estimators require to be

possible to implement varies significantly depending on the algorithm. An
interesting result from the simulation is that the low-complexity LMMSE 2D
method, that requires more information about the channel and also is more complex
than the LMMSE 1D estimator, performs about the same.
Further studies that can be of interest
• A general problem with many of the channel estimation methods that were
analysed in this thesis is that they have a rather high complexity and it
would be interesting to compare their performance to simpler methods as
for example a least squares method with some kind of sliding average as
interpolator.
• Since the interpolation method has such a great impact on the symbol
error, it would be interesting to investigate different interpolation methods
more thorough.
• To study more complex geometrical figures of frame allocations and not

only rectangles in different sizes which is the case in this master thesis.
• Sensitivity of incorrect estimation of SNR for LMMSE 1D and low-

complexity LMMSE 2D
• Sensitivity of incorrect estimation of the Doppler frequency for the low-

complexity Kalman and low-complexity LMMSE 2D
References
[1] IEEE Standard for Local and metropolitan area networks, “Corrigendum 1 Part
16: Air Interface for Fixed Broadband Wireless Access Systems, IEEE Std
802.16e-2005”, 28 Feb 2006
[2] IEEE Standard for Local and metropolitan area networks, “Part 16: Air
Interface for Fixed Broadband Wireless Access Systems, IEEE Std 802.16-2004”,
1 Oct 2004
[3] Fredrik Hansson, “Channel estimation and Interpolation in Multi Antenna

OFDM Based System”, IR-SB-EX-0316, 2003, Royal Institute of Technology
[4] Ove Edfors, Magnus Sandell, Jan-Jaap van de Beek, Sarah Kate Wilson and Per
Ola Börjesson, “OFDM Channel Estimation by Singular Value Decomposition”,
IEEE TRANSACTION ON COMMUNICATIONS, VOL 46, NO 7, July 1998
[5] Hui Lui and Guoqing Li, “OFDM-Based Broadband Wireless Networks Design
and Optimization”, ISBN 0-471-72346-0, John Wiles & Sons, Inc., 2005
[6] W.C. Jakes, “Microwave Mobile Communications”, Wiley, New York, NY,
1974
[7] Way Hong He and Yumin Lee, “Low-Complexity Channel Estimator Structures
for OFDM Systems With and Without Virtual Carriers”, IEEE Communications
Society 0-7803-8533-0/04/$20.00, 2004
[8] Carin Omurcali, “Channel estimation and its effect on the performance in an
OFDM system”, IR-SB-EX-0418, pp. 34-35, Royal Institute of Technology, May
2004
[9] Ye (Geoffrey) Li, Leonard J. Cimini and Nelson R. Sollenberger, “Robust

Channel Estimation for OFDM Systems with Rapid Dispersive Fading Channels”,
IEEE TRANSACTION ON COMMUNICATIONS, VOL 46, NO 7, July 1998
[10] Torsten Söderström and Petre Stocia, “SYSTEM IDENTIFICATION”, ISBN 0-

13-881236-5, pp. 442-443, Prentice Hall International (UK), 1989
84
REFERENCES 85
[11] “Programs for Digital Signal Processing”, IEEE Press, New York 1979,
chapter 8, algorithm 8.1
[12] R.W. Chang and R.A. Gibby, “Theoretical Study of Performance of an

Orthogonal Multiplexing Data Transmission Scheme”, IEEE Transactions on
Communications, 16, 4, pp. 529-540, 1968
[13] J.G. Proakis and Masoud Salehi, “Communication System Engineering second
edition”, Prentice-Hall, Inc., Upper Saddle River, New Jersey, 2002
[14] WiMAX FORUM, “Licensing in the 2 570 – 2 620 MHz band in Sweden”, 29
November, 2005
[15] WiMAX FORUM, “Mobile WiMAX – Part I: A Technical Overview and

Performance Evaluation”, March, 2006
[16] Ove Edfors, Magnus Sandell, Jan-Jaap van de Beek, Daniel Landström and
Frank Sjöberg, “An introduction to orthogonal frequency-division multiplexing”,
Techn. Report Luleå University o Technology, 1996
[17] Eugene Crozier and Allan Klein, “NLOS Optimization Outfits WiMax for Wide
Coverage”, RTC Magazine Inc., March, 2005

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Master of Science Thesis

CHAPTER 1 INTRODUCTION ................................................................. 17

CHAPTER 2 SYSTEM MODEL................................................................. 20

CHAPTER 3 CHANNEL ESTIMATION ................................................... 39

CHAPTER 5 CONCLUSIONS AND FUTURE WORK ............................ 81

The primary focus on IEEE 802.16-2004 is on fixed point-to-point communication,

1.4 Previous work

2.1 Basic principles of OFDM

Figure 2.1: OFDM spectra

where c(k ) is the transmitted symbol on subcarrier k and x(n) is an OFDM

2.1.1 Cyclic prefix

Figure 2.2: Remove ISI through guard interval

Figure 2.3: Cyclic Prefix

2.1.2 OFDM in systems

We begin this subsection by describing some aspects that has to be considered

There is often a need to communicate in both directions, i.e. between BS (base

Figure 2.4: Example of OFDM signal pattern

OFDMA (Orthogonal Frequency Division Multiple Access) is a multi-user version

Figure 2.5: Example of an OFDMA communication

2.2 IEEE 802.16e

2.2.1 Frame structure

An example of the frame structure in TDD mode in IEEE 802.16e (802.16-2005) is

Figure 2.7: Uplink Tile

2.2.2 Pilot generation

Figure 2.8: PRBS generator [1]

2.3.1 Transmission model

Figure 2.9: OFDM transmission model

 w00N L wN0( N −1) 

r (n ) = IDFT (d (n )) ⊗ h(n ) + e(n) (2.10)

y (n ) = DFT (IDFT (d (n )) ⊗ h(n ) + e(n )) =

2.3.2 Channel model

In order to do simulations as close to the reality as possible, it is important to have a

A main property of a channel is if it is time-variant or time-invariant. Time-

c(t ) = c r (t ) + jci (t ) , (2.13)

where cr (t ) and ci (t ) are, stationary, statistically independent and real-valued

a(t ) = cr2 (t ) + ci2 (t ) ,

where τ k is the delay of path k and α k (t ) is the corresponding complex amplitude.

The autocorrelation of the different transmission paths α k (t ) is a zero order Bessel

In the OFDM case different subcarriers correspond to different frequencies. Hence,

where σ k2 is the average energy of the k :th path and *

rt (∆t ) = J 0 (2πf d ⋅ ∆t ) , (2.27)

where J 0 is the zero-order Bessel function and f d is the Doppler frequency.

r f [m] = r (m∆f ) , (2.28)

 r f [0] r f* [1] L r *f [N − 1]

where m is each individual subcarrier and N is the total number of subcarriers.

The correlation in time can be obtained in the following manner,

rt [n] = J 0 (2πf d ⋅ T f n) , (2.30)

 rt [0] rt* [1] L rt* [T − 1]

2.3.3 Signal to noise ratio

2.3.4 Simulation settings

Table 2.1: Information about uplink simulation setup

Figure 2.10: Layout of an uplink frame

Figure 2.11: Tile positioning in frame

3.1.1 Least squares channel estimator

hˆ LS = arghˆ min(y − DWN hˆ LS ) H (y − DWN hˆ LS ) , (3.2)

Q = (WNH D H DWN ) −1 , (3.4)

By substituting (3.4) into (3.5), we obtain

In order to estimate all channel coefficients, the coefficients corresponding to the

Figure 3.1: LS implementation

3.1.2 FIR interpolation algorithm

in which Q is given by (3.4).

Ĥ FIR freq = WP Q P WPH D H y

The inverse that is calculated to attain Q p is therefore performed on a matrix with