1

Preface
Chapter 1 OFDM & Why OFDM: Chapter 1 highlights the importance
of OFDM and its applications are also mentioned.Multipath propagation is
discussed and also the fading caused due to multipath propagation.
Chapter 2 OFDM Basics: It explains the basic principles of Orthogonal
Division Multiplexing (OFDM).OFDM symbol generation and orthogonality is
explained in detail.
Chapter 3 Modulation Schemes: PSK, FSK, BPSK,BFSK,QPSK,QAM
modulation schemes are described which are used in OFDM systems.
Chapter 4 OFDM Transceiver: Basic OFDM transceiver is shown in
this chapter which includes Encoding,Srambling,Convolutional code,viterbi
decoder as well as modulation techniques to be used.
Chapter 5 Detection & Synchronization: This chapter covers Timing
Estimation, Frequency synchronization, Energy Detection, Packet detection.
Chapter 6 MIMO Fundamentals: MIMO(Multiple Input Multiple Output)
basics are discussed.MIMO applications are explained which shows how it
reduces the Mutipath propagation effects like fading,rayleigh fading etc. And
this chapter focuses on space division multiplexing systems (SDM) and space
time block codes (STBC).
Chapter 7 MIMO OFDM MIMO OFDM transceiver design is explained in
this portion of our thesis.It also explains how MIMO OFDM minimizes Inter
symbol interference ISI. The PAC V-BLAST Detection and Decoding Block is
discussed briefly. Different parameters are used to explain the SNR values for
the MIMO OFDM systems.
2
Chapter No.1
OFDM & Why OFDM
1.1 Introduction
The telecommunications industry faces the problem of providing telephone
services to rural areas, where the customer base is small, but the cost of
installing a wired phone network is very high. One method of reducing the high
infrastructure cost of a wired system is to use a fixed wireless radio network. The
problem with this is that for rural and urban areas, large cell sizes are required to
obtain sufficient coverage. This result in problems cased by large signal path loss
and long delay times in multipath signal propagation.
Currently Global System for Mobile telecommunications (GSM) technology is
being applied to fixed wireless phone systems in rural areas.
However, GSM uses Time Division Multiple Access (TDMA), which has a high symbol
rate leading to problems with multipath causing inter-symbol interference.
Several techniques are under consideration for the next generation of digital
phone systems, with the aim of improving cell capacity, multipath immunity, and
flexibility. These include Code Division Multiple Access (CDMA) and Coded
Orthogonal Frequency Division Multiplexing (COFDM). Both these techniques could be
applied to providing a fixed wireless system for rural areas. However, each
technique has different properties, making it more suited for specific applications.
COFDM is currently being used in several new radio broadcast systems
including the proposal for high definition digital television, Digital Video Broadcasting
(DVB) and Digital Audio Broadcasting (DAB). However, little research has been
done into the use of COFDM as a transmission method for mobile telecommunications
systems.
With CDMA systems, all users transmit in the same frequency band using
specialized codes as a basis of channelization. The transmitted information is
spread in bandwidth by multiplying it by a wide bandwidth pseudo random sequence. Both
the base station and the mobile station know these random codes that are used to
modulate the data sent, allowing it to de-scramble the received signal.
OFDM/COFDM allows many users to transmit in an allocated band, by
subdividing the available bandwidth into many narrow bandwidth carriers. Each user is
allocated several carriers in which to transmit their data. The transmission is
generated in such a way that the carriers used are orthogonal to one another, thus
allowing them to be packed together much closer than standard frequency
division multiplexing (FDM). This leads to OFDM/COFDM providing a high spectral
efficiency.
3
1.2 Attenuation
Attenuation is the drop in the signal power when transmitting from one point to
another. It can be caused by the transmission path length, obstructions in the signal path,
and multipath effects. Figure 1.1 shows some of the radio propagation effects
that cause attenuation. Any objects that obstruct the line of sight signal from the
transmitter to the receiver can cause attenuation.Shadowing of the signal can occur
whenever there is an obstruction between the transmitter and receiver. It is generally
caused by buildings and hills, and is the most important environmental attenuation
factor.
Figure1.1 Multi-path Propagation
4
1.3 Multipath Effects
1.3.1 Rayleigh Fading
In a radio link, the RF signal from the transmitter may be reflected from objects such as
hills, buildings, or vehicles. This gives rise to multiple transmission paths at the
Figure1.2 Multi-path Propagation
receiver. Figure 1.2 show some of the possible ways in which multipath signals
can occur.
The relative phase of multiple reflected signals can cause constructive or destructive
interference at the receiver. This is experienced over very short distances (typically at half
wavelength distances), thus is given the term fast fading. These variations can vary
from 10-30dB over a short distance. Figure 4 shows the level of attenuation that can
occur due to the fading.
1.3.2 Frequency Selective Fading
In any radio transmission, the channel spectral response is not flat. It has dips
or fades in the response due to reflections causing cancellation of certain
frequencies at the receiver. Reflections off near-by objects (e.g. ground, buildings,
trees, etc) can lead to multipath signals of similar signal power as the direct signal. This
can result in deep nulls in the received signal power due to destructive interference.
For narrow bandwidth transmissions if the null in the frequency response occurs
at the transmission frequency then the entire signal can be lost. This can be
partly overcome in two ways.
By transmitting a wide bandwidth signal or spread spectrum as CDMA, any dips
in the spectrum only result in a small loss of signal power, rather than a complete loss.
Another method is to split the transmission up into many small bandwidth
carriers, as is done in a COFDM/OFDM transmission. The original signal is
spread over a wide bandwidth and so nulls in the spectrum are likely to only affect a
small number of carriers rather than the entire signal. The information in the lost
carriers can be recovered by using forward error correction techniques.
5
1.3.3 Delay Spread
The received radio signal from a transmitter consists of typically a direct signal, plus
reflections off objects such as buildings, mountings, and other structures. The
reflected signals arrive at a later time then the direct signal because of the extra path
length, giving rise to a slightly different arrival times, spreading the received energy in
time. Delay spread is the time spread between the arrival of the first and last
significant multipath signal seen by the receiver.
In a digital system, the delay spread can lead to inter-symbol interference. This
is due to the delayed multipath signal overlapping with the following symbols. This can
cause significant errors in high bit rate systems, especially when using time division
multiplexing (TDMA). Figure 5 shows the effect of inter-symbol interference due
to delay spread on the received signal. As the transmitted bit rate is increased
the amount of inter-symbol interference also increases. The effect starts to become
very significant when the delay spread is greater then ~50% of the bit time.
Figure 1.3 Multipath Effects
6
Table 8 shows the typical delay spread for various environments. The maximum
delay spread in an outdoor environment is approximately 20ms, thus significant
inter symbol interference can occur at bit rates as low as 25 kbps.
Environment or
cause
Delay Spread Maximum Path Length
Difference
Indoor (room) 40ns 200 ns 12 m 60 m
Outdoor 1 ms 20 ms 300 m 6 km
Table Typical Delay Spread
Inter-symbol interference can be minimized in several ways. One method is to
reduce the symbol rate by reducing the data rate for each channel (i.e. split the
bandwidth into more channels using frequency division multiplexing, or OFDM).
Another is to use a coding scheme that is tolerant to inter-symbol interference
such as CDMA.
1.3.4 Doppler Shift
When a wave source and a receiver are moving relative to one another the
frequency of the received signal will not be the same as the source. When they
are moving toward each other the frequency of the received signal is higher
then the source, and when they are approaching each other the frequency decreases.
This is called the Doppler effect. An example of this is the change of pitch in a cars horn
as it approaches then passes by. This effect becomes important when
developing mobile radio systems.
The amount the frequency changes due to the Doppler effect depends on the relative
motion between the source and receiver and on the speed of propagation of the
wave.
1.4 Frequency Division Multiple Access
For systems using Frequency Division Multiple Access (FDMA), the available
bandwidth is subdivided into a number of narrower band channels. Each user is
allocated a unique frequency band in which to transmit and receive on. During a call, no
other user can use the same frequency band. Each user is allocated a forward
link channel (from the base station to the mobile phone) and a reverse channel (back to the
base station), each being a single way link. The transmitted signal on each of the
channels is continuous allowing analog transmissions.
The channel bandwidth used in most FDMA systems is typically low (30kHz) as each
channel only needs to support a single user. FDMA is used as the primary
subdivision of large allocated frequency bands and is used as part of most multi-
channel systems.
7
Figure 1.4 and Figure 1.5 shows the allocation of the available bandwidth into
several channels.
Figure 1. 4 FDMA showing that the each narrow band channel is allocated to a
single user
Figure 1.5 FDMA spectrum, where the available bandwidth is subdivided into
narrower band channels
8
1.5 Time Division Multiple Access
Time Division Multiple Access (TDMA) divides the available spectrum into
multiple time slots, by giving each user a time slot in which they can transmit or
receive. Figure 1.6 shows how the time slots are provided to users in a round
robin fashion, with each user being allotted one time slot per frame.
Figure 1.6 TDMA scheme where each user is allocated a small time slot
TDMA systems transmit data in a buffer and burst method, thus the transmission of each
channel is non-continuous. The input data to be transmitted is buffered over the previous
frame and burst transmitted at a higher rate during the time slot for the channel.
TDMA can not send analog signals directly due to the buffering required, thus
are only used for transmitting digital data. TDMA can suffer from multipath effects as
the transmission rate is generally very high, resulting in significant inter-symbol
interference.
TDMA is normally used in conjunction with FDMA to subdivide the total
available bandwidth into several channels. This is done to reduce the number
of users per channel allowing a lower data rate to be used. This helps reduce
the effect of delay spread on the transmission. Figure 1.7 shows the use of
TDMA with FDMA. Each channel based on FDMA, is further subdivided using
TDMA, so that several users can transmit of the one channel. This type of
transmission technique is used by most digital second generation mobile phone
systems. For GSM, the total allocated bandwidth of 25MHz is divided into 125,
200 kHz channels using FDMA. These channels are then subdivided further by
using TDMA so that each 200 kHz channel allows 8-16 users .
Figure 1.7 TDMA / FDMA hybrid, showing that the bandwidth is split into 1.3.1
Frequency Division Multiple Access
9
For systems using Frequency Division Multiple Access (FDMA), the available
bandwidth is subdivided into a number of narrower band channels. Each user is
allocated a unique frequency band in which to transmit and receive on. During a call, no
other user can use the same frequency band. Each user is allocated a forward
link channel (from the base station to the mobile phone) and a reverse channel (back to the
base station), each being a single way link. The transmitted signal on each of the
channels is continuous allowing analog transmissions. The channel bandwidth
used in most FDMA systems is typically low (30kHz) as each channel only needs to
support a single user. FDMA is used as the primary subdivision of large
allocated frequency bands and is used as part of most multi-channel systems.
Figure 1.8 and Figure 1.9 shows the allocation of the available bandwidth into
several channels.
Figure 1.8 FDMA showing that the each narrow band channel is allocated to a
single user
Figure 1.9 FDMA spectrums, where the available bandwidth is subdivided into
narrower band channels
10
Figure 1.10 Code division multiple access (CDMA)
1.6 Code Division Multiple Access
Code Division Multiple Access (CDMA) is a spread spectrum technique that
uses neither frequency channels nor time slots. With CDMA, the narrow band
message (typically digitized voice data) is multiplied by a large bandwidth signal
that is a pseudo random noise code (PN code). All users in a CDMA system
use the same frequency band and transmit simultaneously. The transmitted signal is
recovered by correlating the received signal with the PN code used by the
transmitter. Figure 1.10 shows the general use of the spectrum using CDMA.
CDMA technology was originally developed by the military during World War II.
Researchers were spurred into looking at ways of communicating that would be
secure and work in the presence of jamming. Some of the properties that have made
CDMA useful are:
Signal hiding and non-interference with existing systems.
Anti-jam and interference rejection
Information security
Accurate Ranging
Multiple User Access
Multipath tolerance
For many years, spread spectrum technology was considered solely for military
applications. However, with rapid developments in LSI and VLSI designs,
commercial systems are starting to be used.
11
1.7 ADVANTAGES of OFDM
u Efficient way to deal ISI.
u It can better cope with selective fading.
u It uses the spectrum more efficiently.
1.8 APPLICATIONS OF OFDM
1) Digital Audio Broadcasting (DAB)
2) Digital Video Broadcasting (DVB)
3) Wireless Communication
1.8.1DIGITAL AUDIO BROADCASTING
DAB was the first commercial use of OFDM technology [19], [20].
Development of DAB started in 1987 and services began in U.K and Sweden
in 1995. DAB is a replacement for FM audio broadcasting, by providing high
quality digital audio and information services. OFDM was used for DAB due to
its multipath tolerance.
Broadcast systems operate with potentially very long transmission distances
(20 - 100 km). As a result, multipath is a major problem as it causes extensive
ghosting of the transmission. This ghosting causes Inter-Symbol Interference
(ISI), blurring the time domain signal.
For single carrier transmissions the effects of ISI are normally mitigated using
adaptive equalization. This process uses adaptive filtering to approximate the
impulse response of the radio channel. An inverse channel response filter is
then used to recombine the blurred copies of the symbol bits. This process is
however complex and slow due to the locking time of the adaptive equalizer.
Additionally it becomes increasing difficult to equalize signals that suffer ISI of
more than a couple of symbol periods.
OFDM overcomes the effects of multipath by breaking the signal into many
narrow bandwidth carriers. This results in a low symbol rate reducing the
amount of ISI. In addition to this, a guard period is added to the start of each
symbol, removing the effects of ISI for multipath signals delayed less than the
guard period. The high tolerance to multipath makes OFDM more suited to
high data transmissions in terrestrial environments than single carrier transmissions.
Parameter
I
Transmission Mode
II III
IV
Bandwidth 1.536 1.536 MHz 1.536 1.536 MHz
Modulation DQPSK DQPSK DQPS
K
DQPSK
Frequency Range 375 1.5 GHz 3 1.5 GHz
(Mobile reception)
Number of subcarriers
1536 384 192 768
Symbol Duration 1000
ms
250 ms 125 ms 500 ms
Guard Duration 246 ms 62 ms 31 ms 123 ms
Total Symbol Duration 1246
ms
312 ms 156 ms 623 ms
Maximum Transmitter 96 km 24 km 12 km 48 km
Separation for SFN
12
Table 1-1, DAB Transmission parameters for each transmission mode
Table 1-1 shows the system parameters for DAB. DAB has four transmission
modes. The transmission frequency, receiver velocity and required multipath
tolerance all determine the most suitable transmission mode to use.
Doppler spread is caused by rapid changes in the channel response due to
movement of the receiver through a multipath environment. It results in
random frequency
modulation of the OFDM sub carriers, leading to signal degradation. The amount of
Doppler spread is proportional to the transmission frequency and the velocity of
movement. The closer the sub carriers are spaced together, the more susceptible
the OFDM signal is to Doppler spread, and so the different transmission modes in
DAB allow trade off between the amount of multipath protection (length of the guard
period) and the Doppler spread tolerance.
The high multipath tolerance of OFDM allows the use of a Single Frequency
Network (SFN), which uses transmission repeaters to provide improved coverage,
and spectral efficiency. For traditional FM broadcasting, neighboring cities must use
different RF frequencies even for the same radio station, to prevent multipath causes
by re-broadcasting at the same frequency. However, with DAB it is possible for the
same signal to be broadcast from every area requiring coverage, eliminating the
need for different frequencies to be used in neighboring areas.
The data throughput of DAB varies from 0.6 - 1.8 Mbps depending on the amount of
Forward Error Correction (FEC) applied. This data payload allows multiple channels
to be broadcast as part of the one transmission ensemble. The number of audio
channels is variable depending on the quality of the audio and the amount of FEC
used to protect the signal. For telephone quality audio (24 kbps) up to 64 audio
channels can be provided, while for CD quality audio (256 kb/s), with maximum
protection, three channels are available.
More information on DAB can be found in [20] and [21].
1.8.2 DIGITAL VIDEO BROADCASTING
The development of the Digital Video Broadcasting (DVB) standards was started in
1993 . DVB is a transmission scheme based on the MPEG-2 standard, as a method
for point to multipoint delivery of high quality compressed digital audio and video. It is
an enhanced replacement of the analogue television broadcast standard, as DVB
provides a flexible transmission medium for delivery of video, audio and data
services. The DVB standards specify the delivery mechanism for a wide range of
applications, including satellite TV (DVB-S), cable systems (DVB-C) and terrestrial
transmissions (DVB-T). The physical layer of each of these standards. is optimized
for the transmission channel being used. Satellite broadcasts use a single
carrier transmission, with QPSK modulation, which is optimized for this
application as a single carrier allows for large Doppler shifts, and QPSK allows
for maximum energy efficiency [16]. This transmission method is however
unsuitable for terrestrial transmissions as multipath severely degrades the
performance of high-speed single carrier transmissions.
For this reason, OFDM was used for the terrestrial transmission standard for
DVB. The physical layer of the DVB-T transmission is similar to DAB, in that
the OFDM transmission uses a large number of sub carriers to mitigate the
effects of multipath. DVB-T allows for two transmission modes depending on
the number of sub carriers used. Table 1-2 shows the basic transmission
parameters for these two modes. The major difference between DAB and
13
DVB-T is the larger bandwidth used and the use of higher modulation
schemes to achieve a higher data throughput. The DVB-T allows for three sub
carrier modulation schemes: QPSK, 16-QAM (Quadrature Amplitude
Modulation) and 64QAM; and a range of guard period lengths and coding
rates. This allows the robustness of the transmission link to be traded at the
expense of link capacity. Table 1-3 shows the data throughput and required
SNR for some of the transmission combinations.
DVB-T is a uni-directional link due to its broadcast nature. Thus any choice in
data rate verses robustness affects all receivers. If the system goal is to
achieve high reliability, the data rate must be lowered to meet the conditions
of the worst receiver. This effect limits the usefulness of the flexible nature of
the standard. However if these same principles of a flexible transmission rate
are used in bi-directional communications, the data rate can be maximized
based on the current radio conditions. Additionally for multi-user applications,
it can be optimized for individual remote transceivers
Parameter 2k
Mode
8k
Mode
Number sub carriers 1705 6817
Useful Symbol Duration (T
u
) 896 ms 224 ms
Carrier Spacing (1/ T
u
) 1116 Hz 4464 Hz
Bandwidth 7.61
MHz
7.61
MHz
DVB transmission parameters.
Subcarr Code Rate SNR for BER = 2 10
-4
Bit rate (Mbps)
Modulat Viterbi (dB) Guard Period (Fraction of
Useful symbol duration)
Gaussian
Channel
Rayleigh
Channel
1/4 1/32
QPSK 3.1 5.4 4.98 6.03
QPSK 7/8 7.7 16.3 8.71 10.56
16-QAM 8.8 11.2 9.95 12.06
16-QAM7/8 13. 22.8 17.42 21.11
64-QAM 14. 16.0 14.93 18.10
64-QAM7/8 20. 27.9 26.13 31.67
Table 1-3, SNR required and net bit rate for a selection of the coding and
Modulation combinations for DVB
Note: Code rate can be any of the following values: 1/2, 2/3, 3/4, 5/6,
7/8. The Guard Period duration can be any following values: 1/4, 1/8,
1/16, 1/32.
1.8.3 Wireless Communication of OFDM application
The proposed final application for OFDM is to use it for wireless
communications systems such as cellular mobile phone systems, fixed
wireless phone systems, wireless data links and wireless computer local area
networks. If OFDM is to be used in any of these applications then the
bandwidth used must be sufficiently high to compete with other radio
14
technologies. This section discusses the processing power required to implement
a practical OFDM system.
An OFDM system mainly involves digital signal processing, thus the main focus
of the performance of the system depends on the availability of high performance signal
processing. There are two main ways in which the OFDM signal can be processed,
which are using a general purpose DSP, or by implementing the processing in
hardware using customized ICs.
15
Chapter No. 2
OFDM Basics
2.1 Orthogonal Frequency Division Multiplexing (OFDM)
Orthogonal Frequency Division Multiplexing (OFDM) is a multicarrier
transmission technique, which divides the available spectrum into many carriers,
each one being modulated by a low rate data stream. OFDM is similar to FDMA
in that the multiple user access is achieved by subdividing the available
bandwidth into multiple channels, which are then allocated to users. However,
OFDM uses the spectrum much more efficiently by spacing the channels much
closer together. This is achieved by making all the carriers orthogonal to one
another, preventing interference between the closely spaced carriers.
Coded Orthogonal Frequency Division Multiplexing (COFDM) is the same as OFDM
except that forward error correction is applied to the signal before transmission. This is
to overcome errors in the transmission due to lost carriers from frequency
selective fading, channel noise and other propagation effects. For this discussion the
terms OFDM and COFDM are used interchangeably, as the main focus of this thesis is
on OFDM, but it is assumed that any practical system will use forward error
correction, thus would be COFDM.
In FDMA each user is typically allocated a single channel, which is used to transmit all the
user information. The bandwidth of each channel is typically 10 kHz-30 kHz for voice
communications. However, the minimum required bandwidth for speech is only
3 kHz. The allocated bandwidth is made wider then the minimum amount
required to prevent channels from interfering with one another. This extra
bandwidth is to allow for signals from neighboring channels to be filtered out,
and to allow for any drift in the centre frequency of the transmitter or receiver. In a typical
system up to 50% of the total spectrum is wasted due to the extra spacing
between channels. This problem becomes worse as the channel bandwidth becomes
narrower, and the frequency band increases. Most digital phone systems use
vocoders to compress the digitized speech. This allows for an increased system
capacity due to a reduction in the bandwidth required for each user. Current vocoders
require a data rate somewhere between 4-13kbps [13], with depending on the quality
of the sound and the type used. Thus each user only requires a minimum
bandwidth of somewhere between 2-7kHz, using QPSK modulation. However,
simple FDMA does not handle such narrow bandwidths very efficiently.
TDMA partly overcomes this problem by using wider bandwidth channels, which are
used by several users. Multiple users access the same channel by transmitting
in their data in time slots. Thus, many low data rate users can be combined together to
transmit in a single channel that has a bandwidth sufficient so that the spectrum can be
used efficiently.
16
There are however, two main problems with TDMA. There is an overhead associated
with the change over between users due to time slotting on the channel. A
change over time must be allocated to allow for any tolerance in the start time of each
user, due to propagation delay variations and synchronization errors. This limits
the number of users that can be sent efficiently in each channel. In addition, the symbol
rate of each channel is high (as the channel handles the information from
multiple users) resulting in problems with multipath delay spread.
OFDM overcomes most of the problems with both FDMA and TDMA. OFDM
splits the available bandwidth into many narrow band channels (typically 100-
8000). The carriers for each channel are made orthogonal to one another,
allowing them to be spaced very close together, with no overhead as in the FDMA
example. Because of this there is no great need for users to be time multiplex as in TDMA,
thus there is no overhead associated with switching between users.The orthogonality
of the carriers means that each carrier has an integer number of cycles over a
symbol period. Due to this, the spectrum of each carrier has a null at the centre
frequency of each of the other carriers in the system. This results in no
interference between the carriers, allowing then to be spaced as close as
theoretically possible. This overcomes the problem of overhead carrier spacing
required in FDMA.
Each carrier in an OFDM signal has a very narrow bandwidth (i.e. 1 kHz), thus
the resulting symbol rate is low. This results in the signal having a high tolerance
to multipath delay spread, as the delay spread must be very long to cause
significant inter-symbol interference (e.g. > 100 ms).
2.2 Basic Principle of OFDM
The basic principle of OFDM (Orthogonal Frequency Division Multiplexing) is
to split a high-rate data stream into a number of lower rate streams that are
transmitted simultaneously over a number of sub carriers. Because the symbol
duration increases for the lower rate parallel sub carriers, the relative amount
of dispersion in time caused by multipath delay spread is decreased. Inter
symbol interference (ISI) is eliminated almost completely by introducing a
guard time in every OFDM symbol. In the guard time, the OFDM symbol is
cyclically extended to avoid inter carrier interference. This whole process of
generating an OFDM signal and the reasoning behind it are explained in the
following sections.
In OFDM system design, a number of parameters are up for consideration,
such as the number of sub carriers, guard time, symbol duration, sub carrier
spacing, modulation type per sub carrier, and the type of forward error
correction coding. The choice of parameters is influenced by system
requirements such as available bandwidth, required bit rate, tolerable delay
spread, and Doppler values. Some requirements are conflicting. For instance,
to get a good delay spread tolerance, a large number of subcarriers with small
subcarrier spacing is desirable, but the opposite is true for a good tolerance
against Doppler spread and phase noise.
17
Orthogonal Frequency Division Multiplexing (OFDM) is very similar to the well
known and used technique of Frequency Division Multiplexing (FDM). OFDM
uses the principles of FDM to allow multiple messages to be sent over a single
radio channel. It is however in a much more controlled manner, allowing an
improved spectral efficiency.
A simple example of FDM is the use of different frequencies for each FM
(Frequency Modulation) radio stations. All stations transmit at the same time
but do not interfere with each other because they transmit using different
carrier frequencies. Additionally they are bandwidth limited and are spaced
sufficiently far apart in frequency so that their transmitted signals do not
overlap in the frequency domain. At the receiver, each signal is individually
received by using a frequency tuneable band pass filter to selectively remove
all the signals except for the station of interest. This filtered signal can then be
demodulated to recover the original transmitted information.
OFDM is different from FDM in several ways. In conventional broadcasting
each radio station transmits on a different frequency, effectively using FDM to
maintain a separation between the stations. There is however no coordination
or synchronization between each of these stations. With an OFDM
transmission such as DAB, the information signal from multiple stations is
combined into a single multiplexed stream of data. This data is then
transmitted using an OFDM ensemble that is made up from a dense packing
of many subcarriers. All the subcarriers within the OFDM signal are time and
frequency synchronized to each other, allowing the interference between
subcarriers to be carefully controlled. These multiple subcarriers overlap in the
frequency domain, but do not cause Inter-Carrier Interference (ICI) due to the
orthogonal nature of the modulation. Typically with FDM the transmission
signals need to have a large frequency guard-band between channels to
prevent interference. This lowers the overall spectral efficiency. However with
OFDM the orthogonal packing of the subcarriers greatly reduces this guard
band, improving the spectral efficiency.
All wireless communication systems use a modulation scheme to map the
information signal to a form that can be effectively transmitted over the
communications channel
2.3 Generation of subcarriers using IFFT
An OFDM signal consists of a sum of subcarriers that are modulated by using
phase shift keying (PSK) or quadrature amplitude modulation (QAM). If d
i
are
the complex QAM symbols, N
s
is the number of subcarriers, T the symbol
duration, and f
c
the carrier frequency, the one OFDM symbol starting at t = t
s
can be written as:
N
s
/2 -1
s(t) = Re _ d
I + Ns/ 2
exp ( j2 ( f
c
i+0.5/T )(t t
s
)) t
s
< t < t
s
+T
t = N
s/2
s(t) = 0, t < t
s
and t > t
s
+ T (1.1)
In the literature, often the equivalent complex baseband notation is used,
which is given by equation (1.2). In this representation, the real and imaginary
parts correspond to the in-phase and quardrature parts of the OFDM signals,
which have to be multiplied by a cosine and sine of the desired carrier
18
frequency to produce the final OFDM signal. Figure 2.1 shows the operation of
the OFDM modulator in a block diagram.
N
s
/2 -1
s(t) = Re _ d
i+Ns+/ 2
exp ( j2 i/T ( f
c
i+0.5/T )(t t
s
)) t
s
< t < t
s
+T
t = N
s/2
s(t) = 0, t < t
s
and t > t
s
+ T
(1.2)
serial
to
parallel
QAM data
Figure 2.1 OFDM Modulator
As an example, Figure 2.2 shows two subcarriers from one OFDM signal. In
this example, all subcarriers have the same phase and amplitude, but in
practice the amplitudes and phases may be modulated differently for each
subcarrier. Note that each subcarrier has exactly an integer number of cycles
in the interval T, and the number of cycles between adjacent subcarriers
differs by exactly one. This property accounts for the orthogonality between
the subcarriers. For instance, if the j
th
subcarrier from 1.1 is demodulated by
down converting the signal with a frequency of j/T and then integrating the
signal over T seconds, the result is that a complex carrier is integrated over T
i
seconds. For the demodulated subcarrier j, this integration gives the desired
output d
j+N/2
(multiplied by a constant factor T), which is the QAM value for that
particular subcarrier. For all other subcarrier, the integration is zero, because
the frequency difference (i-j)/T produce an integer number of cycles within the
integration interval T, such that the integration result is always zero.
exp(-jN
s
(t-t
s
)/T)
exp(j(N
s
-2)(t-t
s
)/T)
19
Figure 2.2 Example of Four Subcarriers with OFDM
The orthogonality of the different OFDM subcarriers can also be demonstrated
in another way. Each OFDM symbol contains subcarriers that are nonzero
over a T-second period and zero otherwise. The amplitude spectrum of the
square pulse is equal to sinc(pfT), which has zeros for all frequencies f that
are an integer multiple of 1/T. This effect is shown in figure 1.3, which shows
the overlapping sinc spectra of individual subcarriers. At the maximum of each
subcarrier spectrum, all other subcarrier spectra are zero. Because an OFDM
receiver essentially calculates the spectrum values at those points that
correspond to the maxima of individual subcarriers, it can demodulate each
subcarrier free from any interference from the other subcarriers. Basically,
figure 2.3 shows that the OFDM spectrum fulfills Nyquists criterion for an
intersymbol interference free pulse shape. Notice that the pulse shape is
present in the frequency domain and not in the time domain, for which the
Nyquist criterion usually is applied. Therefore, instead of intersymbol
interference (ISI), it is intercarrier interference (ICI) that is avoided by having
the maximum of one subcarrier of one subcarrier spectrum corresponding to
zero crossings of all the others.
Figure 2.3 Spectra of individual subcarriers
20
The complex baseband OFDM signal as defined by equation 1.2 is in fact
nothing more than the inverse Fourier transform of N
s
QAM input symbols.
The tie discrete equivalent is the inverse discrete Fourier transform (IDFT),
which is given by equation:
N
s
-1
s(n) = _ d
i
exp(j2 n/N)
i = 0
where the time t is replaced by a sample n. In practice, this transform can be
implemented very efficiently by the inverse fast Fourier transform (IFFT). An N
point IDFT requires a total of N
2
complex multiplications ----- which are
actually only phase rotations. Of course, there are also additions necessary to
do an IDFT, but since the hardware complexity of an adder is significantly
lower that that of a multiplier or phase rotator, only the multiplications are used
her for comparison. The IFFT drastically reduces the amount of calculation by
exploiting the regularity of the operations in the IDFT. Using the radix-2
algorithm, an N-point IFFT requires only (N/2)-log
2
(N) complex multiplications.
For a 16-point transform, for instance, the difference is 256 multiplications for
the IDFT versus 32 for the IFFT --- a reduction by a factor of 8!. This
difference grows for larger numbers of subcarriers, as the IDFT complexity
grows quadratically with N, while the IFFT complexity only grows slightly
faster than linearly.
The number of multiplications in IFFT can be further reduced by using radix-4
algorithm. This technique makes use of the fact that in a four-point IFFT, there
are only multiplications by {1,-1,j,-j}, which actually do not need to be
implemented by a full multiplier, but rather by a simple add or subtract and a
switch of real and imaginary parts in the case of multiplications by j or j.
2.3.1 ORTHOGONALITY
Signals are orthogonal if they are mutually independent of each other.
Orthogonality is a property that allows multiple information signals to be
transmitted perfectly over a common channel and detected, without
interference. Loss of orthogonality results in blurring between these
information signals and degradation in communications. Many common
multiplexing schemes are inherently orthogonal. Time Division Multiplexing
(TDM) allows transmission of multiple information signals over a single
channel by assigning unique time slots to each separate information signal.
During each time slot only the signal from a single source is transmitted
preventing any interference between the multiple information sources.
Because of this TDM is orthogonal in nature. In the frequency domain most
FDM systems are orthogonal as each of the separate transmission signals are
well spaced out in frequency preventing interference. Although these methods
are orthogonal the term OFDM has been reserved for a special form of FDM.
The subcarriers in an OFDM signal are spaced as close as is theoretically
possible while maintain orthogonality between them.
OFDM achieves orthogonality in the frequency domain by allocating each of
the separate information signals onto different subcarriers. OFDM signals are
made up from a sum of sinusoids, with each corresponding to a subcarrier.
The baseband frequency of each subcarrier is chosen to be an integer
21
multiple of the inverse of the symbol time, resulting in all subcarriers having an
integer number of cycles per symbol.
2.3.2 FREQUENCY DOMAIN ORTHOGONALITY
Another way to view the orthogonality property of OFDM signals is to look at
its spectrum. In the frequency domain each OFDM subcarrier has a sinc,
sin(x)/x, frequency response, as shown in Figure 2-2. This is a result of the
symbol time corresponding to the inverse of the carrier spacing. Each carrier
has a peak at the centre frequency and nulls evenly spaced with a frequency
gap equal to the carrier spacing.
The orthogonal nature of the transmission is a result of the peak of each subcarrier
corresponding to the nulls of all other subcarriers. When this signal is detected using
a Discrete Fourier Transform (DFT) the spectrum is not continuous as shown in
Figure 2.4 (a), but has discrete samples. The sampled spectrum is shown as os in
the figure. If the DFT is time synchronized, the frequency samples of the DFT
correspond to just the peaks of the subcarriers, thus the overlapping frequency
region between subcarriers does not affect the receiver. The measured peaks
correspond to the nulls for all other subcarriers, resulting in orthogonality between
the subcarriers.
(a) (b)
Figure 2.4, Frequency response of the subcarriers in a 5 tone OFDM
signal. (script 0006)
(a) shows the spectrum of each carrier, and the discrete frequency
samples seen by an OFDM receiver. Note, each carrier is sinc,
sin(x)/x, in shape. (b) Shows the overall combined response of the 5
subcarriers (thick black line).
2.3.3 OFDM GENERATION AND RECEPTION
OFDM signals are typically generated digitally due to the difficulty in creating large
banks of phase lock oscillators and receivers in the analog domain. Figure 2-3
shows the block diagram of a typical OFDM transceiver. The transmitter section
converts digital data to be transmitted, into a mapping of subcarrier amplitude and
phase. It then transforms this spectral representation of the data into the time
domain using an Inverse Discrete Fourier Transform (IDFT). The Inverse Fast
Fourier Transform (IFFT) performs the same operations as an IDFT.
22
Chapter No. 3
Modulation Schemes
3.1 Signal Encoding Criteria
Digital signal is a sequence of discrete, discontinuous voltage pulses. Each
pulse is a single element. Binary data is transmitted by encoding each data bit
into single elements. Duration or length of a bit is the amount of time needed
to transmit that bit, and data rate is bits per second.
Now if data rate is increased then Bit Error Rate (BER) will also increase.
However BER will decrease if Signal-to- Noise Ratio (SNR) decreases. There
is another way to improve performance and that is encoding scheme. The
encoding scheme is simply the mapping from data bits to signal elements. A
variety of approaches are in use. However there are various ways to evaluate
these schemes
3.1.1 Signal Spectrum
Several aspects of the signal spectrum are important. A lack of high-frequency
components means that less bandwidth is required for transmission.
Interference depends upon the spectral properties. Usually transfer function of
the channel is worst near the band edges. So good design should concentrate
the transmitted power in the middle of the bandwidth so that distortion be less.
3.1.2 Clocking
Receiver must determine in the beginning and end of each bit position.
Clocking is expensive and hard to achieve as transmitter and receiver must be
synchronized.
3.1.3 Signal interference and noise immunity
Some signal performs much better in the presence of noise.
3.1.4 Cost and complexity
Although digital logic continues to drop in price, this factor should not be
ignored. In particular, the higher the signaling rate to achieve a given data,
greater the cost.
23
3.2 Digital Modulation Schemes
There are three basic encoding or modulation techniques for transforming
digital data into analog signals.
Amplitude Shift Keying (ASK)
Frequency Shift Keying (FSK)
Phase Shift Keying (PSK)
3.2.1 Amplitude Shift Keying
Amplitude shift keying (ASK) in the context of digital communications is a
modulation process, which imparts digital data as variations in the amplitude
of a carrier wave. These are related to the number of levels adopted by the
digital message.
For a binary message sequence there are two levels, one of which is typically
zero. Thus the modulated waveform consists of bursts of a sinusoid.
Figure 3.1 illustrates a binary ASK signal (lower), together with the binary
sequence which initiated it (upper). Neither signal has been band-limited.
Figure 3.1: an ASK signal (below) and the message (above)
There are sharp discontinuities shown at the transition points. These result in
the signal having an unnecessarily wide bandwidth. Band-limiting is generally
introduced before transmission, in which case these discontinuities would be
rounded off. The band-limiting may be applied to the digital message, or the
modulated signal itself. The data rate is often made a sub-multiple of the
carrier frequency. This has been done in the waveform of Figure 3.1.
One of the disadvantages of ASK, compared with FSK and PSK, for example,
is that it has not got a constant envelope. This makes its processing (e.g.,
power amplification) more difficult, since linearity becomes an important
factor. However, it does make for ease of demodulation with an envelope
detector.
The simplest and most common form of ASK operates as a switch, using the
presence of a carrier wave to indicate a binary one and its absence to indicate
a binary zero. This type of modulation is called on-off keying.
24
Figure 3.2 ASK Generation Method
Figure 3.3 shows the signals present in a model of Figure 3.2, where the
message has been band-limited. The shape, after band-limiting, depends
naturally enough upon the amplitude and phase characteristics of the band-
limiting filter.
Figure 3.3: original TTL message (lower), band-limited message (center),
and ASK (above)
It is apparent from Figures 3.1 and 3.4 that the ASK signal has a well defined
envelope. Thus it is amenable to demodulation by an envelope detector. With
band-limiting of the transmitted ASK neither of these demodulation methods
(envelope detection or synchronous demodulation) would recover the original
binary sequence; instead, their outputs would be a band-limited version. Thus
further processing - by some sort of decision-making circuitry for example -
would be necessary. Thus demodulation is a two-stage process:
recovery of the band-limited bit stream
regeneration of the binary bit stream
25
Figure 3.4 ASK Demodulation
The above explanation was for binary modulation. More sophisticated
encoding schemes have been developed which represent data in groups
using additional amplitude levels. For instance, a four-level encoding scheme
can represent two bits with each shift in amplitude; an eight-level scheme can
represent three bits; and so on. These forms of amplitude-shift keying require
a high signal-to-noise ratio for their recovery, as by their nature much of the
signal is transmitted at reduced power.
If L different symbols are to be sent, L different levels of amplitude will be
necessary to achieve the communication. If the maximum amplitude of the
carrier wave is A (with peak-to-peak amplitude of 2A), putting the symbols at
the same distance one from the other, this distance will be:
It is possible to show that the probability to make an error (i.e. a symbol is
read that is different from the one that was sent) is:
where is the complementary error function, GT is the total gain of the
system and oN is the standard deviation of the noise. This relationship is valid
when there is no inter-symbolic interference.
3.2.2 Frequency Shift Keying
As its name suggests, a frequency shift keyed transmitter has its frequency
shifted by the message.
There can be more than two frequencies involved in an FSK signal. The word
keyed suggests that the message is of the on-off (mark-space) variety, such
as one (historically) generated by a Morse key, or more likely in the present
context, a binary sequence. The output from such a generator is illustrated in
Figure 3.5.
26
Figure 3.5: FSK Waveform
Conceptually, and in fact, the transmitter could consist of two oscillators (on
frequencies f1 and f2), with only one being connected to the output at any one
time. This is shown in block diagram form in Figure 3.6
Figure 3.6: FSK Transmitter
Now in FSK Modulators for Binary FSK
s1 = Acos(2[f1t + 1 ) kT < t < (k+1)T for 1
s2 = Acos(2[f2t + 2 ) kT < t < (k+1)T for 0
Where 1 and 2 are the initial phases at t = 0 and T is the bit period of the
binary data. These two signals are not coherent since the phases are not the
same in general. The waveform is not continuous at the bit transitions. This
form of FSK is called non-coherent or discontinuous FSK.
Unless there are special relationships between the two oscillator frequencies
and the bit clock there will be abrupt phase discontinuities of the output
waveform during transitions of the message.
Practice is for the tones s1 and s2 to bear special inter-relationships, and to
be integer multiples of the bit rate. This leads to the possibility of continuous
phase, which offers advantages, especially with respect to bandwidth control.
Alternatively the frequency of a single oscillator (VCO) can be switched
between two values, thus guaranteeing continuous phase - CPFSK. The
continuous phase advantage of the VCO is not accompanied by an ability to
ensure that s1 and s2 are integer multiples of the bit rate. Such type of FSK is
called Coherent FSK. In coherent FSK s1 and s2 are of initially of same phase
at t = 0.
27
Figure 3.7 Coherent FSK modulator
s1 = Acos(2[f1t + ) kT < t < (k+1)T for 1
s2 = Acos(2[f2t + ) kT < t < (k+1)T for 0
This would be difficult to implement with a VCO. FSK signals can be
generated at baseband, and transmitted over telephone lines (for example). In
this case, both s1 and s2 (of Figure 3.6) would be audio frequencies.
For coherent demodulation of the coherent FSK signal, the two frequencies
are so chosen that the two signals are orthogonal:
This leads to
Thus we conclude that for orthogonality fl and f2 must be integer multiples of
1/4T and their difference must be integer multiple of 1/2T. In terms if
frequency difference we have
28
When the separation is chosen as 1/T, then the phase continuity will be
maintained at bit transitions, the FSK called Sunde's FSK, is an important
form of FSK. A particular form of FSK called the minimum shift keying (MSK)
not only has the minimum separation but also has continuous phase. The
figure 3.8 (a) below is of the Sunde's FSK however a coherent FSK might
have discontinuities are bit boundaries, as shown in figure 3.8 (b) below.
Figure 3.8: FSK Waveform
For two non-coherent FSK signals to be orthogonal, the two frequencies must
be integer multiple of 1/2T and their separation must be multiple of 1/T. This
leads to
When n=1, the separation is 1/T, which is minimum. Comparing with coherent
FSK, the separation of non-coherent FSK doubles that of FSK. Thus more
system bandwidth is required for non-coherent FSK for the same symbol rate.
In M-ary FSK, the binary bit stream is divided into n-tuples of n = log2M bits.
We denote all M possible n-tuples as M messages mi, where i = 1,2.M.
There are M signals with different frequencies to represent these M
messages. The expression of the ith signal is
Where T is the symbol rate which is n times the bit rate
29
If the initial phases are the same for all i, the signal set is coherent. As in the
binary case we can always assume the phase to be 0 for coherent MFSK. The
demodulation could be coherent or non-coherent. Otherwise the signal set is
non-coherent and the demodulation must be non-coherent. Usually a uniform
frequency separation between two adjacent frequencies is chosen for MFSK.
3.2.3 Phase Shift Keying
In PSK, the phase of the carrier signal is shifted to represent data.
Any digital modulation scheme uses a finite number of distinct signals to
represent digital data. In the case of PSK, a finite number of phases are used.
Each of these phases is assigned a unique pattern of binary bits. Usually,
each phase encodes an equal number of bits. Each pattern of bits forms the
symbol that is represented by the particular phase.
A convenient way to represent PSK schemes is on a constellation diagram. In
Constellation diagram the real and imaginary axes are often called the in
phase, or I-axis and the quadrature, or Q-axis. By choosing a set of complex
numbers to represent the modulation symbols in this way, they may be
physically transmitted by varying the amplitude of a cosine wave and a sine
wave since these are naturally 90 out-of-phase with one another and are a
convenient representation of the two axes. The amplitude of the cosine-wave
is set to the absolute value of the imaginary part of the symbol to be
transmitted and the amplitude of the sine wave to the absolute value of the
real part.
3.2.4 Binary PSK
BPSK is the simplest form of PSK. It uses two phases which are separated by
180 and so can also be termed 2-PSK. It does not particularly matter exactly
where the constellation points are positioned, and in this figure they are
shown on the real axis, at 0 and 180. This modulation is the most robust of
all the PSKs since it takes serious distortion to make the demodulator reach
an incorrect decision. It is, however, only able to modulate at 1bit/symbol (as
seen in the figure) and so is unsuitable for high data-rate applications.
Figure 3.9: Constellation diagram for BPSK.
30
The bit error rate (BER) of BPSK in AWGN can be calculated as:
Where
Eb = Energy-per-bit
N0 = Noise power spectral density (W/Hz)
Q(x) will give the probability that x is under the tail of the Gaussian probability
density function towards positive infinity
Since there is only one bit per symbol, this is also the symbol error rate. In the
presence of an arbitrary phase-shift introduced by the communications
channel, the demodulator is unable to tell which constellation point is which.
As a result, the data is often differentially encoded prior to modulation.
3.3 Quadrature Phase-shift Keying (QPSK)
More efficient use of bandwidth can be achieved if each signaling element
represents more than one bit. For example, instead of phase shift of 180, as
allowed in PSK, a common encoding technique, know as Quadrature Phase
Shift Keying (QPSK), uses phase shift of multiples of [/2(90).
Among all the MPSK schemes, QPSK is the most often used scheme since it
does not suffer from BER degradation while the bandwidth efficiency is
increased. Other MPSK schemes increase bandwidth efficiency at the
expenses of BER performance.
Since QPSK is a special case of MPSK, its signals are defined as
Where
The initial signal phases are . The carrier frequency is chosen
as integer multiple of the symbol rate, therefore in any symbol interval
[kT,(k+l)T], the signal initial phase is also one of the four phases. The above
expression can be written as
Where 1 (t) and 2 (t) are defined previously.
31
3.3.1 QPSK signal in the time domain
The modulated signal is shown below for a short segment of a random binary
data-stream. The two carrier waves are a cosine wave and a sine wave, as
indicated by the signal-space analysis above. Here, the odd-numbered bits
have been assigned to the in-phase component and the even-numbered bits
to the quadrature component (taking the first bit as number 1).
The total signal the sum of the two components is shown at the bottom.
Jumps in phase can be seen as the PSK changes the phase on each
component at the start of each bit-period. The topmost waveform alone
matches the description given for BPSK above.
Figure 3.10: Timing diagram for QPSK.
The binary data stream is shown beneath the time axis. The two signal
components with their bit assignments are shown the top and the total,
combined signal at the bottom. Note the abrupt changes in phase at some of
the bit-period boundaries.
The binary data that is conveyed by this waveform is: 1 1 0 0 0 1 1 0.
The odd bits, highlighted here, contribute to the in-phase component: 1
1 0 0 0 1 1 0
The even bits, highlighted here, contribute to the quadrature-phase
component: 1 1 0 0 0 1 1 0
3.3.2 Applications of PSK
The most popular wireless LAN standard, IEEE 802.11b, uses a variety of
different PSKs depending on the data-rate required. At the basic-rate of 1
Mbit/s, it uses DBPSK. To provide the extended-rate of 2 Mbit/s, DQPSK is
used. In reaching 5.5 Mbit/s and the full-rate of 11 Mbit/s, QPSK is employed,
but has to be coupled with complementary code keying. The higher-speed
wireless LAN standard, IEEE 802.11g has eight data rates: 6, 9, 12, 18, 24,
36, 48 and 54 Mbit/s. The 6 and 9 Mbit/s modes use BPSK. The 12 and 18
Mbit/s modes use QPSK.
32
The recently-standardized Bluetooth uses /4 DQPSK at its lower rate (2
Mbit/s) and 8-DPSK at its higher rate (3 Mbit/s) when the link between the two
devices is sufficiently robust. Bluetooth 1 modulates with Gaussian minimum-
shift keying, a binary scheme, so either modulation choice in version 2 will
yield a higher data-rate.
3.4 Quadrature Amplitude Modulation
FSK, PSK, CPM and MHPM are constant envelope schemes. The constant
envelope property of these schemes is especially important to systems with
power amplifiers which must operate in the nonlinear region of the input-
output characteristic for the maximum power efficiency like the satellite
transponders. For some other communication systems constant envelope
may not be a crucial requirement whereas bandwidth efficiency is more
important.
Quadrature amplitude modulation (QAM) is such a class of non-constant
envelope schemes that can achieve higher bandwidth efficiency that MPSK
with the same average signal power. QAM is widely used in modems
designed for telephone channels. The CCITT telephone circuit modem
standards V.29 to V.33 are all based on QAM schemes ranging from the un-
coded QAM to the trellis coded 128-QAM. The research of QAM applications
in satellite systems, point-to-point wireless systems and mobile cellular
telephone systems also has been very active.
Ask can also be made M-ary amplitude modulation (MAM). MAM is usually
not used because of its poor power efficiency. However, since QAM signals
consist of two MAM components and they can be demodulated in two
separate channels. It is necessary to understand QAM.
3.4.1 QAM Signal Description
The description of the MAM signal given above was necessary to understand
QAM. In MAM schemes the signals have the same phase but different
amplitudes. In MPSK schemes, signals have the same amplitudes but
different phases. Naturally, the next step of development is to consider both
amplitude and phase modulations in a scheme (QAM). That is
Where Ai is the amplitude and 8j is the phase of the ith signal in the M-ary
signal set. Pulse shaping is usually used to improve the spectrum and for ISI
control purpose in QAM. With pulse shaping, the QAM signal is
Where p(t)is a smooth pulse in the interval [O,T].We can write the above
equation as
33
Where
And
Similar to MPSK, QAM signal can be expressed as a linear combination of
two ortho-normal functions. Expression is
Where
And
Where Ep is the energy of p(t)in [O,T].Basically the basis functions are ortho-
normal and when there is no pulse shaping they are perfectly ortho-normal.
The energy of the ith signal is
And the average signal energy is
34
The average power is
The average amplitude is
3.4.2 QAM Constellation
The first QAM scheme was presented by c.R.Cahn in 1960. He simply
extended phase modulation to a multi-amplitude phase modulation. That is
there is more than one amplitude associated with an allowed phase. In the
constellation a fixed number of signal points (or phasors) are equally spaced
on each of the N circles, where N is the number of amplitude levels. This is
called type I constellation. In this type, the points on the inner ring are closest
together in distance and are most vulnerable to errors. To overcome this
problem, type II constellation was proposed by Hancock and Lucky a few
months later. In this type, the signal points are still on circles but the number
of points on the inner circle is less than the number of points on the outer
circle, making the distance between two adjacent points on the inner circle
approx. equal to that on the outer circle. Type III constellation is the square
QAM constellation which was proposed by Campopiano and Glazer in 1962.
Their analysis showed that this system offered a very small improvement in
performance over the type II system but the implementation would be
considerably simpler than that of both the types. Due to this type III
constellation has been the most widely used system. Some of the other two
dimensional constellation is shown in the figure below:
Figure 3.11 QAM III Constellation
35
3.4.3 Rectangular QAM
Rectangular QAM constellations are, in general, sub-optimal in the sense that
they do not maximally space the constellation points for a given energy.
However, they have the considerable advantage that they may be easily
transmitted as two pulse amplitude modulation (PAM) signals on quadrature
carriers, and can be easily demodulated. The non-square constellations, dealt
with below, achieve marginally better bit-error rate (BER) but are harder to
modulate and demodulate.
The first rectangular QAM constellation usually encountered is 16-QAM, the
constellation diagram for which is shown here. A Gray coded bit-assignment is
also given. The reason that 16-QAM is usually the first is that a brief
consideration reveals that 2-QAM and 4-QAM are in fact binary phase-shift
keying (BPSK) and quadrature phase-shift keying (QPSK), respectively. 8-
QAM presents problems in dividing an odd number of bits between the two
carriers, and is rarely used since 8-PSK is considerably simpler.
Figure3.12 Constellation diagram for rectangular 16-QAM.
3.13 Constellation diagram 3.14 Alternative constellation
for rectangular diagram for rectangular 8-QAM
8-QAM.
36
Chapter No. 4
OFDM Transceiver
4.1 BLOCK DIAGRAM OF OFDM TRANSCEIVER
37
4.2 SCRAMBLER
It is electronic instrument which make the speech unintelligible during
transmission and restores it at reception. It is also used to remove strings of
zeros & ones. Military makes it's first used.
Figure 4.3 Scrambler
4.3 ENCODER
It is Forward Error Correction (FEC) technique. Encoding Is Used In Order To
Encrypt. The Data So That Only Intended User Can Read It .Convolutional
Encoder with constraint length 7 is generally used.
.
Figure 4.4 Encoder
4.4 Convolutional Codes
A convolutional code maps each k bits of a continuous input stream on n
output bits where the mapping is performed by convolving the input bits with a
binary impulse response. The convolutional encoding can be implemented by
simple shift registers and modulo-2 adders. As an example, figure 2.1 shows
the encoder for a rate cod which is actually one of the most frequently
38
applied convolutional codes. This encoder has a single data input and two
ouputs A
i
and B
i
, which are interleaved to form the coded output sequence
{ A
1
,B
1
,A
2
,B
2
}. Each pair of output bits { A
i
,B
i
} depends on seven input bits,
being the current input bit plus six previous input bits that are stored in the
length 6 shift register. This valued of 7 or in general the shift register length
plus 1 is called the constraint length. The shift register taps are often
specified by the corresponding generator polynomials or generator vectors.
Figure 4.5 Convolutional encoder (k = 7)
Decoding of convoltional codes is most often performed by soft decision
Viterbi decoding, which is an efficient way to obtain the optimal maximum
likelihood estimate of the encoded sequence. Simpler versions of decoding of
convolutional codes are hard decision Viterbi decoding & sequential decoding.
The description of Viterbi decoding will be done in the next section.
The complexity of Viterbi decoding grows exponentially with the constraint
length. Hence, practical implementations do not go further than a constraint
length of about 10. Decoding of convolutional codes with larger constraint
length is possible by using suboptimal decoding techniques like sequential
decoding.
Because convolutional codes do not have a fixed length, it is more difficult to
specify their performance in terms of Hamming distance and a number of
correctable errors. One measure that is used is the free distance, which is the
minimum Hamming distance between arbitrarily long different code sequences
that begin and end with the registers of the encoder. For example, the code of
figure 2.1 has a free distance of 10. When hard decision decoding is used this
cod can correct up to floor ((10-1)/2) = 4 bit errors within each group of
encoded bits with a length of about 3 to 5 times the constraint length. When
soft decision decoding is used, however, the number of correctable errors
does not really give a useful measure anymore. A better performance
measure is the coding gain, which is defined as the gain in the bit energy-to-
noise density ratio E
b
/N
o
relative to an uncoded system to achieve a certain bit
error ratio. The E
b
/N
o
gain is equivalent to the gain in input signal-to-noise
ration (SNR) minus the rate loss in dB because of the redundant bits.
A convolutional code can be punctured to increase the coding rate. For
instance, increasing the rate of the above rate code to is done by deleting
2 of every 6 bits at the output of the encoder. The punctured output sequence
for a rate code is { A
1
B
1
A
2
B
3
A
4
B
4
A
5
B
6
A
7
B
7
}. For a rate 2/3 code, the
39
punctured output sequence is { A
1
B
1
A
2
A
3
B
3
A
4
A
5
B
5
}. To decode the
punctured sequence, the original rate decoder can be used. Before
decoding, erasures have to be inserted in the data at the locations of the
punctured bits.
4.5 Viterbi decoding
The decoder used for decoding convolutional codes is Viterbi decoder. Viterbi
decoder can be of two types: soft decision viterbi decoding or hard decision
viterbi decoding. Here hard decision viterbi decoding is explained.
To understand decoding process, it simplifies matters to expand the state
diagram to show the time sequence of the encoder. Let's take an example of
the following convolutional encoder and its state diagram.
Figure 4.6 Convolutional Encoder with ( n , k , K ) = ( 2 , 1 , 3 )
if the state diagram is laid out vertically then the expanded diagram, called a
trellis, is constructed by reproducing the states horizontally and showing the
state transitions going from left to right corresponding to time, or data input. If
the constraint length K is large, then the trellis diagram becomes unwieldy. In
that case, 2
K-2
simplified trellis fragments can be used to depict the transitions.
Any valid output is defined by a path through the trellis. In our example a-b-c-
b-d-c-a-a produces the out put 11 10 00 01 01 11 00 and was generated by
the input 1011000. If an invalid path occurs, such as a-c, then the decoder
attempts error correction. In essence, the decoder must determine what data
input was most likely to have produced the invalid output.
Figure 4.7 Trellis Diagram for Encoder of fig.4.4
40
A number of error correction algorithms have been developed for
convolutional codes. Perhaps the most important is the Viterbi code. In
essence, the Viterbi technique compares the received sequence with all
possible transmitted sequences. The algorithm chooses a path through the
trellis whose coded sequence differs from the received sequence in the fewest
number of places. Once a valid path is selected as the correct path, the
decoder can recover the input data bits from the output code bits.
There are several variations on the Viterbi algorithm, depending on which
metric is used to measure differences between received sequences and valid
sequences. To give an idea of the operation of the algorithm, we use th
common metric of Hamming distance. We represent a received coded
sequence as the word w = w
0
w
1
w
2
, and attempt to find the most likely valid
path through the trellis. At each time i and for each state we list the active path
(or paths) through the trellis to the state. An active path is a valid path through
the trellis whose Hamming distance from the received word up to time i is
minimal. We label each state at time i by the distance of its active path from
the received word. The following relationship is used:
distance of a path = distance of the last edge + distance of the last-but-one
state
The algorithm proceeds in b+1 steps, where b is pre chosen window size. For
an ( n,k,K ) code, the first output block of n bits, w
o
w
1
w
2
w
n
, is decoded in the
following steps:
Step 0:
The initial state of the trellis at time 0 is labeled 0, because there is so
far no discrepancy.
Step i +1:
For each state S at time i + 1, find all active paths leading to S using
the above equation. Label S by the distance of that path or paths.
Step b:
The algorithm terminates at time b. If all the active paths at that time
have the same first edge and the label of that edge is x
0
x
1
x
2
x
n-1
, then
the first code block w
0
w
1
w
2
w
n-1
is corrected to x
0
x
1
x
2
x
n-1
. If there
are two active edges, the error is not correctable.
After accepting and, if necessary, correcting, the first code block, the decoding
window is moved n bits to the right and the decoding of the next block is
performed.
Convolutional codes provide good performance in noisy channels where a
high proportion of the bits are in error. Thus, they have found increasing use in
wireless applications.
41
4.6 INTERLEAVING
All encoded data bits shall be interleaved by a block interleaver with a block
size corresponding to the number of bits in a single OFDM symbol, NCBPS.
The interleaver is defined by a two-step permutation.
The first permutation is defined by the rule
i = (NCBPS/16) (k mod 16) + floor(k/16) k = 0,1,,NCBPS 1
The second permutation is defined by the rule
j = s floor (i/s) + (i + NCBPS floor (16 i/NCBPS)) mod s i = 0, 1,
NCBPS 1
Because of the frequency selective fading of typical radio channels, the OFDM
subcarriers generally have different amplitudes. Deep fades in the frequency
spectrum may cause groups of subcarriers to be less reliable than others,
thereby causing bit errors to occur in bursts rather than being randomly
scattered. Most forward error correcting codes are not designed to deal with
error burst. Therefore, interleaving is applied to randomize the occurrence of
bit errors prior to decoding. At the transmitter, the coded bits are permuted in
a certain way, which makes sure that adjacent bits are separated by several
bits after interleaving. At the receiver, the reverse permutation is performed
before decoding. A commonly used interleaving scheme is the block
interleaver, where input bits are written in a matrix column by column and read
out row by row.
4.7 SUBCARRIER MAPPING
Different types of Mapping technique are used.BPSK,QPSK,16-QAM, 64-
QAM Depending upon the data rate required. Max Data rate of 54Mbits/s can
be achieved using 64-QAM.
The OFDM subcarriers shall be modulated by using BPSK, QPSK, 16-QAM,
or 64-QAM
modulation depending on the RATE requested. We use BPSK MODULATION
in the signal field. For Data field Transmission we used 16-QAM which can
achieve max data rate of 36 Mbits/s with coding rate of 3/4.
d = (I + jQ) * KMOD is the output.
4.7.1 Quardrature amplitude modulation
Quardrature amplitude modulation (QAM) is the most popular combination
with OFDM (other modulation techniques used are: BPSK and QPSK).
Especially rectangular constellations are easy to implement as they can be
split into independent pulse amplitude modulated (PAM) components for both
the in-phase and quardrature part. Figure 4.8 shows the rectangular
constellations of Quradrature Phase Shift Keying (QPSK), 16-QAM and 64-
QAM. The constellations are not normalized; to normalize them to an average
power of one assuming that all constellation points are equally likely to
occur each constellation has to be multiplied by the normalization factor
listed in table 4.1. The table also mentions Binary Phase Shift Keying (BPSK),
which uses two of the four QPSK constellation points { 1+j , -1-j }.
42
Table 4.1 Modulation-dependent normalization factor KMOD
Figure 4.8 BPSK, QPSK, 16-QAM, and 64-QAM constellation bit encoding
43
4.7.2 SUBCARRIER MODULATION
Once each subcarrier has been allocated bits for transmission, they are
mapped using a modulation scheme to a subcarrier amplitude and phase,
which is represented by a complex In-phase and Quadrature-phase (IQ)
vector. Figure 4.9 shows an example of subcarrier modulation mapping. This
example shows 16-QAM, which maps 4 bits for each symbol. Each
combination of the 4 bits of data corresponds to a unique IQ vector, shown as
a dot on the figure. A large number of modulation schemes are available
allowing the number of bits transmitted per carrier per symbol to be varied
Digital data is transferred in an OFDM link by using a modulation scheme on each
subcarrier. A modulation scheme is a mapping of data words to a real (In phase) and
imaginary (Quadrature) constellation, also known as an IQ constellation.
Figure 4.9, Example IQ modulation constellation. 16-QAM, with gray
coding of the data to each location. (script s0045)
4.8 PILOT INSERTION
In each OFDM symbol, four of the subcarriers are dedicated to pilot signals in
order to make the coherent detection robust against frequency offsets and
phase noise. These pilot signals shall be put in subcarriers 21, 7, 7 and 21.
In each OFDM symbol, four of the subcarriers are dedicated to pilot signals in
order to make the coherent detection robust against frequency offsets and
phase noise. These pilot signals shall be put in subcarriers 21, 7, 7 and 21.
The pilots shall be BPSK modulated by a pseudo binary sequence to prevent
the generation of spectral lines. The contribution of the pilot subcarriers to
each OFDM symbol is described in the following section
4.9 IFFT
Inverse Fast Fourier Transform (IFFT) is used to Convert Frequency Domain
Data into Time Domain.
44
4.9.1 FREQUENCY TO TIME DOMAIN CONVERSION
After the sub carrier modulation stage each of the data subcarriers is set to an amplitude and
phase based on the data being sent and the modulation scheme; all unused subcarriers are set to
zero. This sets up the OFDM signal in the frequency domain. An IFFT is then used to convert this
signal to the time domain, allowing it to be transmitted. Figure 6.10 shows the IFFT section of the
OFDM transmitter. In the frequency domain, before applying the IFFT, each of the discrete
samples of the IFFT corresponds to an individual subcarrier. Most of the subcarriers are modulated
with data. The outer subcarriers are unmodulated and set to zero amplitude. These zero
subcarriers provide a frequency guard band before the nyquist frequency and effectively act as an
interpolation of the signal and allows for a realistic roll off in the analog anti-aliasing reconstruction
filters.
Figure 4.10 IFFT section of OFDM
4.9.2 Cyclic Extension/Guard Interval
It is used to avoid Iner Symbol Interference (ISI).Guard Time must be larger than expected
delay spread such that multipath components from one symbol cannot interfere with next
symbol.
One of the most important properties of OFDM transmissions is its high level of robustness
against multipath delay spread. This is a result of the long symbol period used, which minimizes the
inter-symbol interference. The level of multipath robustness can be further increased by the
addition of a guard period between transmitted symbols. The guard period allows time for
multipath signals from the pervious symbol to die away before the information from the
current symbol is gathered. The most effective guard period to use is a cyclic extension of the
symbol. If a mirror in time, of the end of the symbol waveform is put at the start of the symbol as the
guard period, this effectively extends the length of the symbol, while maintaining the
orthogonality of the waveform. Using this cyclic extended symbol the samples required for
performing the FFT (to decode the symbol), can be taken anywhere over the length of the
symbol.
45
For a given system bandwidth the symbol rate for an OFDM signal is much lower than a
single carrier transmission scheme. For example for a single carrier BPSK modulation, the
symbol rate corresponds to the bit rate of the transmission. However for OFDM the system
bandwidth is broken up into N
c
subcarriers, resulting in a symbol rate that is N
c
times lower
than the single carrier transmission. This low symbol rate makes OFDM naturally resistant
to effects of Inter-Symbol Interference (ISI) caused by multipath propagation.
Multipath propagation is caused by the radio transmission signal reflecting off objects in the
propagation environment, such as walls, buildings, mountains, etc. These multiple signals arrive at
the receiver at different times due to the transmission distances being different. This spreads the
symbol boundaries causing energy leakage between them.
The effect of ISI on an OFDM signal can be further improved by the addition of a guard period to
the start of each symbol. This guard period is a cyclic copy that extends the length of the symbol
waveform. Each subcarrier, in the data section of the symbol, (i.e. the OFDM symbol with no guard
period added, which is equal to the length of the IFFT size used to generate the signal) has an
integer number of cycles. Because of this, placing copies of the symbol end-to-end results in a
continuous signal, with no discontinuities at the joins. Thus by copying the end of a symbol and
appending this to the start results in a longer symbol time. Figure 4-11 shows the insertion of a
guard period.
Symbol N
Figure 4.11, Addition of a guard period to an OFDM signal
The total length of the symbol is
Ts=TG
+
TFFT,
where T
s
is the total length of the
symbol in samples, TG is the length of the guard period in samples, and
TFFT
is the size of the IFFT
used to generate the OFDM signal.
In addition to protecting the OFDM from ISI, the guard period also provides protection against time-
offset errors in the receiver.
46
Chapter No. 5
Packet Detection
5.1 Introduction
In this chapter, robust and efficient frame detection and symbol timing synchronization
technique suitable for IEEE 802.11a wireless LAN system is proposed. The proposed
method does frame detection using a threshold comparison mechanism and performs
Orthogonal Frequency Division Multiplexing (OFDM) symbol boundary detection using
correlation techniques. This algorithm is a novel combination of self and cross correlation
information to achieve symbol timing synchronization. The proposed algorithm can robustly
detect the symbol boundary even under low SNRs, high frequency offset, and multipath.
The algorithm proposed for frame detection and symbol timing synchronization is designed
by exploiting the repetitive nature of the short preambles provided in the 802.11a preamble.
Finding the symbol timing for OFDM systems is nothing but finding the beginning of the
OFDM symbol. This can be achieved by finding any boundary in the preamble.
5.2 OFDM-WLAN system
Figure 7.1 shows a simplified transceiver structure of OFDM based 802.11a system. The
OFDM signal can be generated by taking IDFT (Inverse Discrete Fourier Transform) of
QAM or PSK symbols. As per IEEE 802.11a specification, each OFDM symbol consists of
48 data carriers, 4 pilot carriers and 12 null carriers. Hence, IDFT size is 64 point and can
be implemented using efficient IFFT algorithm. The output of IFFT becomes one OFDM
symbol, with duration of Ts (3.2ms). Each OFDM symbol is cyclically extended with 16
samples of duration T
g
(0.8 ms) and they will be removed at the receiver. The cyclic prefix
length should be more than the channel impulse response to avoid Inter Symbol
Interference (ISI). The expression for the OFDM baseband signal (output of IDFT) is:
(1)
Where C
k
is the QAM or PSK modulated complex signal and N = 64. This baseband signal
is then up-converted, modulated to radio frequency (RF) and transmitted. IEEE 802.11a is
a packet-based communication system. Each packet is preceded by a preamble as defined
in IEEE 802.11a specification. The preamble structure is as shown in figure 6.2. This figure
also shows the information regarding signal field and data payload.
As shown in the figure, the preamble consists of 10 short symbols each having 0.8 s
duration, and two long preambles of 3.2 s duration each.
47
Figure 5.1 Frame format and Preamble structure of IEEE 802.11a
5.3 Detection of frame
At receiver, the received signal is correlated with itself with a delay of one short symbol,
given by
.. (2)
Where r(n) is the received sequence, A(n) is the correlation output and L is the length of the
short symbol. The incoming frame at receiver can be detected by comparing the magnitude
of auto-correlation result with some threshold. It is advisable to have a dynamic threshold
based on incoming signal power. The initial 2-3 short symbols are assumed to be non-
reliable, as Automatic Gain Control (AGC) logic requires some time to finalize the gain
setting.
5.4 Symbol boundary detection
In this chapter, a robust algorithm to detect OFDM symbol boundary using auto-correlation
and cross correlation of short preamble is described. In (2) the value of N should be in
between 16 and 144 and a multiple of 16. For any particular value of N, if we plot the auto-
correlation magnitude values, we get a curve as shown in figure 3. The curve rises to some
value, remains flat for about N-CP samples duration and then falls down as shown. In our
algorithm, we detect the index of the (N-CP+1)
th
sample when counted from the start of the
preamble. The auto-correlation magnitude values are passed through a moving average
filter to smoothen the curve. The moving average filter is defined by:
Where A(n) is the auto-correlation magnitude and l is the size of the moving average filter
and it is chosen as 3. The falling edge of the curve corresponds to the (N-CP)th sample.
We can detect this falling edge by observing the slope of the curve. However, at low SNRs
and high delay spread situations, exact detection of this edge is difficult. This edge can be
localized with the help of cross correlation of the received sequence. If we do cross
correlation of the received sequence with the local copy of the short symbol, we get peaks
at the end of each short symbol. However, the frequency offset of the local oscillator
disturbs the magnitude of these cross correlation peaks significantly. Instead of averaging
48
this cross correlation for one short symbol, if we average over more short symbols, as
indicated by (4), we can still detect the peak.
Where s(n) is the local copy of the short symbol. M is the number of short symbols over
which we are averaging the cross correlation. The simulation result shows that this type of
averaging can with stand 20ppm frequency offset, which is the worst-case possible
frequency offset specified 802.11a standard.
Figure 5.2 Auto-Correlation curve and Cross-Correlation peaks for ideal case (No noise, no
multipath and no frequency offset)
Figure 5.2 shows the auto-correlation curve and the cross correlation peaks in the case of
an ideal channel. Our objective is to detect the cross correlation peak at which the falling
edge of the auto-correlation curve starts. This can be achieved by tracking the slope of the
curve with the help of another dynamically set threshold. To detect the corresponding cross
correlation peak, we need to do peak search by taking a window of 16 cross correlation
magnitude values around that falling edge. The window size can be decided through
simulations. The detected peak corresponds to the index of the (N-CP+1)th sample when
counted from the start of the preamble. Thus, in the proposed algorithm, cross correlation is
used to localize the exact boundary of the short symbol. The difference between the
detected boundary and the actual boundary is the boundary detection error. This boundary
detection error results in corresponding rotation of the signal constellation in frequency
domain. This rotation can be taken care by channel estimation and equalization up to some
extent.
Robust frame detection and symbol timing detection algorithms were proposed for IEEE
802.11a WLAN system. In particular, a novel combination of auto-correlation and cross-
correlation information was used to increase the reliability of the frame boundary detection
algorithm.
Synchronization
49
5.5 INTRODUCTION
Synchronization is an essential task for any digital communication system. Without any
synchronization algorithms, it is not possible to reliably receive the transmitted data. From
the digital baseband algorithm design engineers perspective, synchronization algorithms
are the major design problems that have to be solved to build a successful product.
OFDM is used for both broadcast type systems and a packet switched networks, like
WLANs. These two systems require somewhat different approach to the synchronization
problem. Broadcast system transmit data continuously, so a typical receiver, like European
Digital Audio Broadcasting (DAB) or Digital Video Broadcasting (DVB) system receivers,
can initially spend a relatively long time to acquire the signal and then switch to tracking
mode. On the other hand, WLAN systems typically have to use so called single-shot
synchronization; that is, the synchronization has to be acquired during a very short time
after the start of the packet. This requirement comes from the packet switched nature of the
WLAN systems and also from the data rates used. To achieve the good system throughput,
it is mandatory to keep the receiver training information overhead to minimum. To facilitate
the single-shot synchronization, current WLAN standards include problems, like IEEE
802.11a preamble shown in figure 3.5 or the various HiperLAN/2 preambles, in the start if
packet. The length and the contents of the preamble have been carefully designed to
provide enough information for good synchronization performance without any unnecessary
overhead.
The current waveform makes most of the synchronization algorithms designed for a single
carrier system unusable, thus algorithm design problem has to be approached from the
OFDM perspectives. This distinction is especially visible on the sensitivity difference to
various synchronization errors between single carrier and OFDM systems. The frequency
domain nature of OFDM also allows the effect of several synchronization errors to be
explained with the aid of the properties of the Discrete Fourier Transform (DFT). Another
main distinction can be performed either in time- or frequency-domain. This flexibility is not
available in single carrier systems. The tradeoffs in how to perform the synchronization
algorithms are usually either higher performance versus reduced computational complexity.
The order of algorithms described in this chapter follows to some extent the order of how
an actual receiver would perform the synchronization.
The main assumption usually made when WLAN systems are designed is that the channel
impulse response does not change significantly during one data burst. This assumption is
justified by the quite short time duration of transmitted packets, usually a couple
milliseconds at maximum and that the transmitter and receiver in most applications move
very slowly relatively to each other. Under this assumption most of the synchronization for
WLAN receivers is done during the preamble and need not be changed during the packet.
5.6 TIMING ESTIMATION
Timing estimation consists of two main tasks: packet synchronization and symbol
synchronization.
The IEEE 802.11 MAC protocol is the essentially a random access network, so the receiver
does not know exactly when a packet starts. The first task of the receiver is to start of an
incoming packet. HiperLAN/2 medium access architecture is a hybrid of both random
access and time scheduled networks. Thus HiperLAN/2 receiver also has to be able to
reliably detect the start of an incoming packet, without prior knowledge.
50
Broadcasting systems naturally do not require packet detection algorithms, because
transmission is always on. However, for packet oriented network architecture, finding the
packets is obviously of central importance for high network performance.
5.7 PACKET DETECTION
Packet detection is the task of finding an approximate estimate of the start of the preamble
of an incoming data packet. As such it is the first synchronization algorithm that is
performed, so the rest of the synchronization process is dependent on good packet
detection performance. Generally packet detection can be described as a binary hypothesis
test. The test consists of two complementary statements about a parameter of interest.
These statements are called the null hypothesis, H
0,
and the alternative hypothesis, H
1
. In
the packet detection test, the hypotheses assert whether a packet is present or not. This
set up as shown below.
H
0
: Packet not present
H
1
: Packet present
The actual test is usually of the form that test is whether decision variable m
n
exceeds a
predefined threshold Th. The packet detection case is shown below.
H
0
: m
n
< Th => Packet not present
H
1
: m
n
> Th => Packet present
The performance of the packet detection algorithm can be summarized with two
probabilities: probability of detection P
D
and probability of false alarm P
FA
. P
D
is the
probability of detecting a packet when it is truly present, thus high P
D
is desirable quality for
the test. P
FA
is the probability that the test incorrectly decides that the packet is present,
thus P
FA
should be as small as possible. In general, increasing P
D
increases P
FA
and vice
versa, hence the algorithm designer must settle for some balanced compromise between
the two conflicting goals.
Generally it can be said that a false alarm is a less severe error than not detecting a data
packet at all. The reason is that after a false alarm, the receiver will synchronize to
nonexistent packet and will detect its error at the first received data integrity check. On the
other hand, not detecting a packet always results in lost data. A false alarm can also result
lost data, in case an actual receiver will not be able to catch the packet. The probability of
this occurring depends on several issues like the network load and the time it takes for the
receiver to detect its mistake. In conclusion, a little higher P
FA
can be tolerated to guarantee
the good P
D
.
5.8 RECEIVED SIGNAL ENERGY DETECTION
The simplest algorithm for finding the start edge of the incoming packet is to measure the
received signal energy. When there is no packet being received, the signal r
n
consists only
of noise r
n
= w
n
. When the packet starts, the received energy is increased by the signal
component r
n
= s
n
+ w
n
, thus the packet can be detected as a change in the received
energy level. The decision variable m
n
is then the received signal energy accumulated over
some window of length L to reduce the sensitivity to large individual nose samples.
L 1 L
m
n
= r
n k
r
*
n k
= | r
n k
|
2
51
k =0
Equation 5.1
Calculation of m
n
can be simplified by nothing that it is moving sum of its received signal
energy. This type of sum is also called a sliding window. The rational for the name is that
the energy time instant n, one new value enters the sum and one old value is discarded.
This structure can be used to simplify the computation of equation 5.1. Equation 5.2 shows
how to calculate the moving sum recursively.
m
n+1
= m
n
+ | r
n+1
|
2
- | r
n-L+1
|
2
Equation 5.2
Thus the number of complex multiplications is reduced to one per received sample;
however, more energy is required to store all the value of |r
n
|
2
inside the window. The
response of this algorithm is show in Figure 3.1. The figure shows the value of m
n
for IEEE
802.11 a packet with 10dB Signal to Noise Ratio (SNR) and sliding window length L = 32.
The true start of the packet is at n = 500, thus in this case the threshold could be set
between 10 and 25. The value of the threshold defines the P
D
and P
FA
of the test. This
simple method suffers from a significant drawback; the value of the threshold depends on
the received signal energy. When the receiver is searching for an incoming packet, the
received signal consists of only noise. The level of the noise power is generally unknown
and can change when the receiver adjusts its Radio Frequency (RF) amplifier setting s or if
unwanted interferers go on and off in the same band as the desired system. When a
wanted packet is incoming, its received signal strength depends on the power setting of the
transmitter and on the total path loss from the transmitter to the receiver. All these factors
make it quite difficult to set a fixed threshold, which could be used to decide when an
incoming packet starts. The next section describes an improvement to the algorithm that
alleviates the threshold value selection problem.
Figure 5.1 Received Energy based packet detection algorithm
5.9 DOUBLE SLIDING WINDOW PACKET DETECTION
The double sliding window packet detection algorithm calculates two consecutive sliding
windows of the received energy. The basic principle is to form the decision variable m
n
as a
ratio of the total energy contained inside the two windows. Figure 2.2, the A and B windows
and response of m
n
to a received packet. In figure 2.2, the A and B windows are
considered stationary relative to the packet that slides over them to the right. It can be seen
52
that when only noise is received the response is flat, since both windows contain ideally the
same amount of noise energy. When the packet edge starts to cover the A window, the
energy in the window A gets higher until the point A is totally contained inside the start of
the packet. This point is the peak of the triangle shaped m
n
and the position of the packet in
Figure 5.2 corresponds to this sample index n. After this point B window starts to also
collect signal energy, and when it is also completely inside the received packet, the
response of m
n
is flat again. Thus the response of m
n
be thought of as a differentiator, in
that its value is large when the input level changes rapidly. The packet detection is declared
when m
n
crosses over the threshold value Th.
Figure 5.2 Double sliding window packet detection
Equation 5.3 shows the calculation of A window value and the equation 5.4 the calculation
for B window.
M 1 M 1
a
n
= r
n m
r
*
n m
= | r
n m
|
2
m = 0 m = 0
Equation 5.3
L L
b
n
= r
n+l
r
*
n+l
= | r
n+l
|
2
l = 1 l = 0
Equation 5.4
Both a
n
and b
n
are again sliding windows, thus computation can be simplified in the same
recursive manner as for the energy detection window. Then the decision variable is formed
by dividing the value of a
n
by b
n
.
m
n
= a
n
/ b
n
Equation 5.5
A simulated response of the variable m
n
is shown in the figure 5.3. The figure is again for
the IEEE 802.11 a preamble with 10dB SNR. The figure clearly shows how the value of m
n
does not depends on the total received power. After the peak, the response levels off to the
same as before the peak, although the received energy level is much higher. An additional
benefit of this approach is that, at the peak point of m
n
the value of a
n
contains the sum of
the signal energy N and the b
n
value is equal to to the noise energy N, thus the value of m
n
at the peak point and Equation 5.6 is the ratio a
n
and b
n
at the peak point and Equation 5.7
is the SNR. Equation 5.6 is the ratio a
n
and b
n
at the peak point and Equation 5.7 is the
SNR estimate calculated from the ratio.
53
m
peak
= a
peak
/ b
peak
= S+N
= S + 1
N
Equation 5.6
^
SNR = m
peak
- 1
Equation 5.7
Using the double sliding window algorithm is a good approach if the receiver does not have
additional information about the received data.
Figure 5.3 Double sliding window packet detection
5.10 FREQUENCY SYNCHRONIZATION
One of the main drawbacks of OFDM is its sensitivity to carrier frequency offset. The
degradation is caused by two main phenomena; reduction of amplitude of the desired
subcarrier and ICI caused by neighboring carrier carriers. The amplitude loss occurs
because the desired subcarrier is no longer sampled at the peak of the sinc-function of
DFT. The sinc-function is defined as sinc (x) = sin (x)/x. Adjacent carriers cause
interference, because they are not sampled at the zero-crossing of they sinc-function. The
overall effect on SNR is analyzed and for relatively small frequency errors, the degradation
in dB was approximated by
SNR = 10 ( T f

)
2
E
s
N
0
N
0
54
where f is the frequency error as a fraction of subcarrier spacing and T is the sampling
period. The performance effect varies strongly with the modulation used; naturally,
constellations with fewer points can tolerate larger frequency errors than large
constellations. Figure 5.4 illustrates the effect for QPSK and 64-QAM constellations.
The figure shows that 64-QAM, the largest constellation in IEEE802.11a, cannot tolerate
more than 1% error in the carrier frequency for negligible SNR loss of 0.5dB, whereas
QPSK modulation can tolerate up to 5% error for the same SNR loss.
The analysis above would seem to indicate that large constellations are exceedingly
difficult to use with an OFDM system. However, keep in mind that a large constellation
automatically implies higher operating SNR than with a small constellation. This directly
improves the performance of the frequency error estimation.
In Hseih and Wei, the various algorithms that have been developed to estimate carrier
frequency offset in OFDM systems are divided in three types.
Figure 5.4 Symbol error rate (SER) degradation due to frequency offset at SER = 10
-4
Type 1
Data-aided algorithms; these methods are based on special training
information embedded into the transmitted signal.
Type 2
Non-data-aided algorithms that analyze the received signal in the
frequency domain.
Type 3
55
Cyclic prefix based algorithms that use the inherit structure of the OFDM signal
provided by the cyclic prefix.
For WLAN applications, type 1 is the most important. The preamble allows the receiver to
use efficient maximum likelihood algorithm to estimate and correct for the frequency offset,
before the actual information portion of the packet starts. The algorithm belonging to type 2
and 3 are better suited for broadcast or continuous transmission type OFDM.
Figure 5.5 The IEEE 802.11a standard preamble
Chapter No. 6
MIMO Fundamentals
6.1 Introduction
The objective of this chapter is to describe and compare the multiple-input
multiple-output (MIMO) techniques which have been proposed in literature in the past
years. Unfortunately most of the available papers focus on narrowband (single carrier)
wireless communication. Few papers consider broadband applications and usually they use
assumptions tailored for mobile communication. The following describes for a number of
narrowband MIMO algorithms and, when available, the broadband adaptation or
application.
Now the focus is on the use of smart antennas as a candidate for next generation
broadband cellular or wireless systems. The use of smart antennas is a potential
breakthrough in the improvement of quality, transmission capacity and data rates in
wireless and cellular systems. Basically, the idea is to make use of a strong multi-path
propagation environment instead of being limited by it. In the receiver, the multi-path
streams are transformed into independent channels by using a combination of adaptive
antennas, appropriate transmission coding and receiver detection algorithms. The hidden
capacity of a wireless medium could be exploited by the introducing spatial (de-
)multiplexing of several communications channels within the same bandwidth and the same
time-slot. This is shown in Figure
56
Figure6.1 Multiple Communication Channels
The use of multiple antennas at both transmit and receive side is also referred to as
multiple-input multiple-output (MIMO) systems. MIMO systems could serve multiple
objectives. This is shown in Table 6-1.
Table 6.1 Several ways to exploit the unbounded air-interface to improve either coverage,
link
quality, capacity or data rates.
Multiple receive antennas will lead to array gain (3 dB signal-to-noise improvement when
doubling the number of receive antennas) even in a non multipath environment. This could
be used to increase the coverage of a base-station or access point. Another possibility is
use interference reduction schemes at the receive array to suppress interference from
unwanted sources and/or users.Basically, this will improve the link quality but also more
users can be used in a cell, which will lead to more capacity. In a multi-path fading
environment, a large improvement (more than 10 dB) can be obtained by using multiple
receive antennas by mitigating the weak signal fades. The focus of this work package will
be on high data rates; therefore the spatial multiplexing is chosen as the suitable candidate.
Spatial multiplexing has the potential to increase the transmission rate with a factor n,
where n is the minimum of the number of transmit and receive antennas.
6.2 MIMO Theoretical Capacity
Before analyzing different MIMO techniques, it is important to show the capacity that can
be achieved by using multiple transmitters and multiple receivers. These theoretical bounds
57
will be used later to compare how close the algorithm capacity is to the theoretical one.
Here, a summary is given of the capacity for different channels as reported in [Fos98].
The following assumptions have been considered to formulate the results given below:
Number of transmit and receive antennas is, respectively, Nt and Nr
The total transmit signal vector is composed of Nt statistically independent
equal power components. The total irradiated power is independent of the
number of transmit antennas.
The communication channel H is assumed to be flat over frequency
Average SNR at each receiver branch: p = P/N0 where P is the received power
independent of Nt
Noise at the receiver is modeled as AWGN vector of dimension Nr, with
independent components of identical power N0 for each of the Nr branches.
The capacity for a MIMO system, for the assumptions given above is:
where (.)H denotes the conjugate transpose of a vector or matrix. Since usually in indoor
environments, we come across channels that can be modeled by a Rayleigh distribution, it
is useful to report the capacity achievable in such channels for different antennas
configurations.
We now assume:
Quasi-static channel H (Nt columns, Nr rows): i.e. the randomly selected
channel is not changing during a single packet transmission.
Channel follows the Rayleigh distribution. The element of the channel H (Nr x Nt matrix)
are i.i.d., complex, zero-mean, unit variance entries:
consequently, |hij|^2 is a ^2 variate but normalized so E(|hij|^2) = 1.
H is known at the receiver.
The capacity for different antennas configurations becomes [Uys01]:
A) No diversity: Nr = Nt = 1
58
B) Receive Diversity: Nt = 1, Nr = N
C) Transmit Diversity: Nt = N, Nr = 1
D) Combined Transmit-Receive Diversity: Nt = Nr
The best theoretical capacity is achieved in case D. This theoretical result confirms that the
combination of transmit and receive diversity improves the data rate and the performance
of wireless links.
6.3 MIMO and SIMO Feasibility Study under Rayleigh
Conditions
6.3.1Introduction
The feasibility study is confined to the prediction of capacities in indoor non lineof-sight
wireless channels. The propagation environment typically consists of many scatterers, so
that the signal travels multiple paths before arriving at the receiver. Such a propagation
environment is shown in Figure 6.2.
Figure 6.2 Indoor Wireless Fading Environment
In the limit of many scatterers, it can be shown that the Rayleigh distribution is
appropriate to describe the spatial fluctuations of the amplitude of a narrowband signal.
Shannons equation explains the dependency of the capacity C on the bandwidth B and
signal-to-noise ratio SNR.
59
Lets take the bandwidth fixed (because of spectral regulations or system
requirements). A Rayleigh number generator is taken for the signal-to-noise ratio and we
study the probability density function (PDF) of the normalized capacity (given by log2
(1+SNR)).
Figure 6.4 Probability density function of the capacity [bit/s-Hz] under Rayleigh fading
conditions
The capacity is treated as a stochastic variable. It is characterized by its mean value (around
5 bits/s-Hz in the figure above) and the variation around the mean. The variation around the
mean has a relation to the reliability of the wireless
communications link. MIMO aims to propose systems with high data rates and a good
reliability. The reliability can be evaluated by having a look at the tail of the PDF. Therefore,
the concept outage is now introduced. Outage is the chance that the user will experience a
capacity below x bits/s-Hz.
In the remainder of this report, outage levels of 10 % and 1 % will be taken.
Drawing a cross-section of the CDF with those outage levels will give us the
capacity. This capacity is guaranteed within 90 % or 99 % of the indoor space,
respectively.
6.3.2 Average capacity of SIMO and MIMO systems
Consider a system with nT transmit antennas and nR receive antennas. The
generalized Shannon capacity for this MIMO system can be written as shown in Eq. (1). A
Rayleigh number generator is used to describe the spatial variations of the signal-to-noise
ratio p. The Rayleigh fading model is the most popular model which describes fading as a
complex Gaussian process. The elements of the channel matrix are taken as unit variance
circular symmetric Gaussian stochastic variables.
The average capacity for various MIMO systems as a function of the average
signal-to-noise ratio F is shown in Figure 6-5. In addition, the capacity for a single input
multiple-output system (SIMO) is also shown. The SIMO case denotes the classical
diversity system with only multiple antennas on one side of the communication link.
60
Figure 6.5
Average capacity of a n = 1...4 SIMO system (oc) compared to (n,n) MIMO system
(orth) for uncorrelated Rayleigh fading channels.
As a reference, the conventional single antenna system (SISO) is included as a reference.
Consider that conventional systems (GSM, UMTS etc..) have rather low spectral
efficiencies around 1 2 bits/s-Hz. Take an arbitrary target of 8 bits/s-Hz for our broadband
MIMO research, the following table shows the signal-to-noise ratio to reach this capacity
target.
Table Signal-to-noise ratio [dB] to get a average spectral efficiency of 8 bits/s-Hz.
6.3.4 Outage capacity of SIMO and MIMO systems
A mean capacity of 8 bit/s-Hz is an interesting figure, but it does not tell us what the
capacity will be 90 % or 99 % of the time or space. Therefore, another interesting
performance measure for the reliability of the service is the outage capacity. Outage
capacity means that the capacity equal or higher than a certain fixed value is guaranteed
61
for 90 % or 99 % of all spatial locations. The SNR threshold for an outage capacity of 8
bit/s-Hz is shown in Table.
Table: Signal to noise ratio to get an outage spectral efficiency of 8 bit/s-Hz for outage of
10 %
and 1 %, respectively.
Note that a conventional single antenna system requires an unrealistic 44 dB
signal-to-noise ratio to achieve a reliable capacity of 8 bits/s-Hz, while a (4,4)
MIMO system only needs a signal-to-noise ratio of 10 dB.
6.4 Multi-path Propagation and MIMO
Multipath propagation is a feature of all wireless communication environments. There is
usually a primary (most direct) path from a transmitter at point A to a receiver at point B.
Inevitably, some of the transmitted signal takes other paths to the receiver, bouncing off
objects, the ground, or layers of the atmosphere.
Signals traversing less direct paths arrive at the receiver later and are often
attenuated. A common strategy for dealing with weaker multipath signals is to
simply ignore themin which case the energy they contain is wasted. The strongest
multipath signals may be too strong to ignore, however, and can degrade the performance
of wireless LAN equipment based on existing standards.
Radio signals can be depicted on a graph with the vertical axis indicating amplitude and the
horizontal axis indicating time as sine waves. See Figure 1, a. When a multipath signal
arrives slightly later than the primary signal, its peaks and troughs are not quite aligned with
those of the primary signal, and the (combined) signal seen by the receiver is somewhat
blurred. See Figure 1, b. If the delay is sufficient to cause the multipath signals peaks to
line up with the primary signals troughs, the multipath signal will partially or totally cancel
out the main signal. See Figure 1, c.
Traditional radio systems either do nothing to combat multipath interference, relying on the
primary signal to out-muscle interfering copies, or they employ multipath mitigation
techniques. One mitigation technique uses multiple antennas to capture the strongest
62
signal at each moment in time. Another technique adds different delays to received signals
to force the peaks and troughs back into alignment. Whatever the mitigation technique, all
assume multipath signals are wasteful and/or harmful and strive to limit the damage.
63
MIMO, in contrast, takes advantage of multipath propagation to increase throughput,
range/coverage, and reliability. Rather than combating multipath signals, MIMO puts
multipath signals to work carrying more information. This is accomplished by sending and
receiving more than one data signal in the same radio channel at the same time. The use
of multiple waveforms constitutes a new type of radio communicationcommunication
using multi-dimensional signalswhich is the only way known to improve all three basic
link performance parameters (range, speed and reliability). Because MIMO transmits
multiple signals across the communications channel (rather than the conventional systems
single signal), MIMO has the ability to multiply capacity (which is another word for speed).
A common measure of wireless capacity is spectral efficiencythe number of units of
information per unit of time per unit of bandwidthusually denoted in bits per second per
Hertz, or b/s/Hz.
Using conventional radio technology, engineers struggle to increase spectral efficiency
incrementally (i.e. one b/s/Hz at a time). By transmitting multiple signals containing different
information streams over the same frequency channel, MIMO provides a means of doubling
or tripling spectral efficiency.
MIMO can also be thought of as a multi-dimensional wireless communications system.
Conventional radio systems try to squeeze as much information as possible through a one-
dimensional pipe. In order to do that, engineers must adapt their designs to the noise and
other limitations of a one-dimensional channel. MIMO empowers engineers to work in
multiple dimensions, creating opportunities to work around the limitations of a one-
dimensional channel.
Greater spectral efficiency translates into higher data rates, greater range, an increased
number of users, enhanced reliability, or any combination of the preceding factors. By
multiplying spectral efficiency, MIMO opens the door to a variety of new applications and
enables more cost-effective implementation for existing applications.
An interesting sidelight: Guglielmo Marconi demonstrated the first non line-of-sight (NLOS)
wireless communications system in 1896 by communicating over a hill. From that day
forward, engineers viewed multipath signals as an annoyance at best and serious problem
at worst. The first paper describing wireless MIMOs capacity multiplying capability was
published 100 years later in the 1996 Global Communications Conference proceedings.
How Does MIMO Differ from the Smart Antenna?
MIMO and smart antenna systems may look the same on first examination: Both employ
multiple antennas spaced as far apart as practical. But look under the hood, and you will
64
see that MIMO and smart antenna systems are fundamentally different. Smart antennas
enhance conventional, one-dimensional radio systems. The most common smart antenna
systems use beamforming (a.k.a. beam switching) to concentrate the signal energy on the
main path and receive combining (a.k.a. diversity) to capture the strongest signal at any
given moment. Note that beamforming and receive combining are only multipath mitigation
techniques, and do not multiply data throughput over the wireless channel. See Figure 2.
Thats not to say beamforming and receive combining arent useful. Both have
demonstrated an ability to improve performance incrementally in point-to-point
applications (e.g., outdoor wireless backhaul applications). However, while
beamforming and receive combining are valuable enhancements to conventionalradio
systems,
MIMO is
a
paradig
m shift,
dramatic
ally
changing
perceptio
ns of and
response
s to
multipath
propagat
ion.
While
receive
combinin
g and
beamfor
ming
increase
spectral
efficienc
y one or
two
b/s/Hz at
a time;
MIMO
multiplie
s the
b/s/Hz.
6.5
MIMO
IEEE
Standa
rd
802.11
n
65
MIMO-OFDM technology is more than the latest technical improvement for wireless LANs.
MIMO-OFDM is a major technology upgrade enabling demanding new applications with
huge market potential and facilitating significant growth in existing applications. Though the
IEEE 802.11n task group is developing a standard for MIMO-OFDM wireless LAN devices
around which the entire industry should rally, the delivery of pre-standard MIMO enhanced
Wi-Fi devices today can only boost development of a robust market for MIMO-OFDM
wireless LAN devices.
MIMO Algorithms
6.6 MIMO Algorithms and Architectures
Different MIMO architectures have been proposed in literature of the last few
years. This chapter reviews the different MIMO techniques. It focuses on space division
multiplexing systems (SDM) and space time block codes (STBC).
6.7 Space Division Multiplexing and Layered Architectures
SDM techniques exploit the spatial dimension using multiple antennas at both
transmitter and receiver. These techniques transmit different signals on different transmit
antennas simultaneously. The goal is to increase the capacity and the SNR performance.
At the receiver, the different signals are recovered using the Space Division Multiplexing
technique. Multiple antennas are required at the receiver to recover the transmit signals
more accurately.
Figure 6.6 The physical model of a system with SDM.
6.8 Signal Model
A communication system comprising Nt transmit (TX) and Nr receive (RX)
antennas are considered. This system, assumed to operate in a Rayleigh flat-fading
environment, exploits the spatial dimension by using Space Division Multiplexing (SDM)
(see Figure 6.6, where MAPU stands for Multi Antenna Processing Unit). Suppose we
model the channel impulse response H as a zero-mean complex Gaussian variable, like:
H = A + jB (1)
where A and B are zero-mean statistically independent real Gaussian variables, each
having a variance o2/2. The variance of the complex Gaussian variable H can be shown to
be:
66
Suppose that at discrete times, the transmitter sends an Nt-dimensional (complex) signal
vector s (i.e., it transmits Nt parallel streams of data), and the receiver records an Nr-
dimensional complex vector x. Then the following signal model describes the relation
between s and x:
where H is an Nr Nt complex propagation matrix that is constant with respect to the
symbol time and assumed known at the receiver (e.g. via transmitting training sequences)
and the vector n (Nr -dimensional) represents additive receiver noise.
The vector s is assumed to have zero-mean, uncorrelated random variables with variance
equal to o s2. The total power of s (i.e., E[sHs]) is assumed to be Ps. Thus, the covariance
matrix of s equals:
where H denotes the conjugate transpose of a vector or matrix and the matrix I with
subscript Nt represents the identity matrix with dimension Nt. Note that the total transmitted
power does not depend on the number of transmit antennas but is assumed fixed at Ps.
The vector n is Nr -dimensional and represents additive receiver noise. The vector n is
assumed to have zero-mean, uncorrelated random variables with variance o^2 and a
covariance matrix equal to:
Furthermore, it is assumed that the vectors s and n are independent and thus the following
holds: E[snH ] = 0. To explain the different Space Division
Multiplexing techniques, the following notations will be used:
where xi and si represent the i-th element of x and s respectively. The Hi and hi vectors
denote the i-th row and i-th column of H, respectively.
Transmission schemes
Depending on the coding architecture used in the multiple antenna transmitter, two main
types of SDM scheme are possible: per antenna coding and joint coding scheme.
Joint coding:
In this transmission scheme a single code is used to encode all the signals going to
different antennas. After coding, interleaving and mapping, the Serial input bit stream is
converted to Nt Parallel sub-streams (S/P). See figure 6.7.
67
Figure 6.7
6.9 Antenna coding:
In this transmission scheme the serial input bit stream is first converted in Nt parallel sub-
streams, then each sub-stream is separately coded. See figure 6.8.
Figure 6.8 MIMO transmission scheme deploying a per antenna coding architecture.
6.10 Linear receiver scheme
The Zero Forcing Algorithms
The ZF algorithm is based on a conventional adaptive antenna array (AAA)
technique, namely, linear combinatorial nulling [Wol98]. In this technique, each sub-stream
in turn is considered to be the desired signal, and the remaining data streams are
considered as interferers. Nulling of the interferers is performed by linearly weighting the
received signals so that all interfering terms are cancelled.
For Zero Forcing, nulling of the interferers can be performed by choosing weight vectors
di (with i = 1, 2, , Nt) such that
where T stands for the transpose of a vector or matrix and hj denotes the j-th
column of the channel matrix H. However, when we take a closer look to this
criterion, solving the weight vectors is equal to finding a matrix D such that:
where D is a matrix that represents the linear processing in the receiver. The i-th row of D
is equal to the transpose of the i-th weight vector di and I is the identity matrix. So, by
68
forcing the interferers to zero, each desired element of s can be estimated. If H is not
square, D equals the pseudo-inverse of H:
where + represents the pseudo-inverse. In order for the pseudo-inverse to exist, Nt must be
less than or equal to Nr, because for Nt larger than Nr, HHH is singular and its inverse does
not exist [Str88]. Furthermore, note that in order for the inverse to exist, the columns of H
must be independent. Regarding the independent and identically distributed (i.i.d.)
assumption of the elements of H, independence is usually an approximation, which is
justifiable if 1) the antenna spacing is chosen equal to or larger than //2 [Fos98] (where /
represents the wavelength of the transmission frequency) and 2) the system operates in a
rich-scattered environment, which can be modeled by Rayleigh flat-fading. Thus, for Nr < Nt
and if the inverse of HHH exists, the estimates of s (given by sest) can be found by:
or, equivalently:
Using Formula (11), (sest)i, i.e. the i-th component of sest, can be written as:
where +Hi represents the i-th row of H+ , which, according to Formula (7), is
equal to the transpose of the i-th weight vector di. Note that di is a so-called nulling vector
[Wol98]. As a final step, (sest)i can be sliced to the nearest Quadrature Amplitude
Modulation (QAM) constellation point, these sliced signals are denoted by s . In this way,
all Nt elements of s can be decoded at the receiver. The diversity order of an (Nt,Nr)
system based on ZF is equal to NrNt+1, as shown in [Win94].
Note that a diversity order of one means that the BER improves by a factor of 101 if the
SNR is increased by 10 dB. In case of a diversity order of two, if the SNR is increased by
10 dB, the BER improves 102 times, etc.
6.11 The Minimum Mean Square Error solution
Another approach in estimation theory to the problem of estimating a random
vector s on the basis of observations x is to choose a function g(x) that minimizes the
Mean Square Error (MSE).
An exact function g(x) is usually hard to obtain, however, is we restrict this
function to be a linear function of the observations, an exact solution can be
achieved. Using linear processing, the estimates of s can be found by:
Now, to obtain the linear Minimum Mean Square Error (MMSE), D must be
chosen such that the Mean Square Error s2 is minimized:
69
To minimize the Mean Square Error (over D), the processing at the receiver must be equal
to:
where d is equal to on^2/ os^2 = Nt /p.
From Formula (10) it becomes clear that the ZF solution correspond to an MMSE solution
with d = 0.
6.12 Non Linear receiver scheme
ZF with Decision Feedback Decoding
The linear nulling approach as described in the previous section is viable, but as will
become clear from the results in Section 4, superior performance is obtained if non-linear
techniques are used. One can imagine that if somehow first the most reliable element of
the transmitted vector s could be decoded and used to improve the decoding of the other
elements of s, superior performance can be achieved. This is called symbol cancellation
[Wol98] and it exploits the timing synchronism inherent in the system model (the
assumption of co-located transmitters makes this completely reasonable). Furthermore,
linear nulling (i.e., ZF) is used to perform detection. In other words, symbol cancellation is
based on the subtraction of
interference from already detected components of s from the receiver signal vector x. This
results in a modified receiver vector in which, effectively, fewer interferers are present.
Because this principle is somewhat analogous to decision feedback equalization, it is also
called Decision Feedback Decoding (DFB).
When symbol cancellation is used, the order in which the components of s are
detected becomes important to the overall performance of the system. To determine a
good ordering of detection, the covariance matrix of the estimation error s sest will be
used. For ZF, this covariance matrix can be shown to be:
or, using the pseudo-inverse:
70
Let (sest)i be the i-th entry of sest, then, the "best" estimate, (sest)i, is the one for which Pii
(i.e., the i-th diagonal element of PH) is the smallest, because this is the estimate with the
smallest error covariance. From Formula (18) it becomes clear that Pii is equal to the
squared length of the i-th row of the pseudo-inverse. So find the minimum squared length
row of H+ is equivalent. Suppose that the order in the pseudo-inverse of H is arranged so
that the row with the least squared length becomes the last row (the i-th row of H+ is
permuted with the Nt-th row), then the Nt-th element of sest can be independently
decoded. Let N s denote the decoded value, then this value can be used to improve the
estimate of the remaining Nt1 signals (i.e., symbol cancellation). If this procedure to find
the best estimate is
performed in a recursive way, the so-called Optimal Detection (OD) method as described in
[Wol98] is obtained. Here it is called the Decision Feedback Decoding algorithm with
optimal detection. The recursive algorithm can be described as follows:
1. Compute H+ ;
2. Find the minimum squared length row of H+ and permute it to be the last row,permute
the columns of H accordingly;
3. Form the estimate of the last component of s. In case of ZF:
where the transpose of +
Nt H is said to be the Nt-th nulling vector [Wol98] ;
4. Obtain s Nt (via slicing) from
5. (While ~1 > 0 t N ) go back to step 1, but now with:
Note that in case step 2 is skipped, the DFB algorithm is performed without
optimal detection and the overall performance will be less, however, processing time is
saved.
6.13 Minimum Mean Square Error with Decision Feedback Decoding (V-
BLAST)
In order to perform Decision Feedback Decoding with Minimum Mean Square
Error decoding, the DFB algorithm of Section 2.6.1 has to be adapted somewhat.
Again, the covariance matrix of the estimation error s sest will be used to
determine a good ordering of detection. For MMSE, this covariance matrix can be shown to
be:
71
Note that P is somewhat different from the case where ZF is used as detection. In order the
do DFB based on the MMSE algorithm, the DFB algorithm is adapted and becomes:
1. Compute D (P is obtained while computing D);
2. Find the smallest diagonal entry of P and suppose this is the i-th entry. Permute the i-th
column of H to be the last column and permute the rows of D accordingly;
3. Form the estimate of the best component of s:
where Nt D represents the last row of D and its transpose is the Nt-th nulling vector
[Wol98];
4. Obtain s Nt (via slicing) from
5. (While ~1 > 0 t N ) go back to step 1, but now with:
In the following chapters, we will refer to this algorithm as V-BLAST (Vertical
Bell Laboratory Space Time Architecture) which is the widely spread name used in
literature to identify this technique.
6.14 Maximum likelihood Decoding
MLD is a method that compares the received signal with all possible transmitted signals
and estimates s according to the Maximum Likelihood principle. Suppose a matrix C gives
all possibilities in s that could occur (the dimensions of C are N K t , where K = QNt and
Q represents the number of constellation points).
Then, the receiver should store a matrix Y such that:
72
At the receiver, the most likely transmitted signal is determined, as the one for
which
is minimal (with 1 > j > K), i.e., the signal sj that corresponds with the vector yj
which lays closest to the received vector is said to be the most likely signal to be
transmitted. Thus, s is chosen to be the j-th column of C. This can be rewritten to the
following formula where sml represents the maximum likelihood detection of the transmitted
signal s:
MLD is optimal in terms of BER performance. However, a major disadvantage is that the
complexity of MLD is proportional to QNt , due to the fact that the size of Y grows
exponentially with Nt .
Another way to show the superiority in BER performance of MLD over the other SDM
techniques is by checking its diversity order. It is shown in [Nee00-1] that the diversity order
of a MLD system with Nr receive antennas is equal to Nr.Note that in the case of MLD, it is
not required that Nt > Nr.
Soft Output MLD
The MLD technique can be modified to deliver not only the most likely transmitted symbol,
but also reliable values, which are known as soft-decision outputs. Hagenauer presents in
[Hag89] a method to derive soft-decision values. There, the log likelihood ratio is used as
an indication for the reliability of a bit. If x denotes the received vector, bl is the l-th bit to
estimate, H is the estimated channel matrix and sj is one of the possible transmitted
vectors (with 1 > j > K, where K = QNt and Q represents the number of constellation
points), then the L-value of the estimated bit is:
Because the vectors sj are equally likely to be transmitted, P(sj) is equal for all vectors sj.
Using the probability density function of a multivariate normal distribution give a certain
channel, we find the soft-output decisions:
73
Soft Output technique will be used together with MLD for simulation. We will
refer to it as SOMLD.
6.15 Space time block codes
In STBC the input to the encoder is a stream of modulated symbols from a real or complex
constellation. The encoder operates on a block of K symbols which are distributed on
different antennas (space) and on T symbol times. The results are matrix code words
whose rows correspond to antennas and columns correspond to symbol times. The ratio
K/T gives the coding rate.
6.15.1 Alamouti scheme
At the transmitter side Alamouti scheme exploits two transmitting antennas that in two
transmission time slots (T1, T2), emit two symbols according to the following scheme
[Ala98]:
Alamouti scheme.
Where the sign * means conjugation.
74
The first stage receiver combines the signals coming from two different transmit branches
in the two consecutive times while the second stage performs Maximum Likelihood
Detection (MLD).
The received signal will be:
The above equations can be written using the equivalent orthogonal channel matrix as:
If the channel is known, the estimated transmit signal can be easily found through the
channel match-filter. This algorithm reaches the optimum capacity [Ala98].
6.15.2Generalization of Alamouti Scheme
The generalization of orthogonal matrix code words for more than 2 transmit
antennas does not reach unitary transmission rate. Different Space Time Blocks schemes
for 3 and 4 antennas were proposed. see Table
Proposed STBC schemes.
75
Conclusions
Different MIMO algorithms are presented in this chapter.
Space Division Multiplexing techniques:
ZF: is linear and does not use any other knowledge that the channel
estimation. Its diversity is equal to Nr-Nt+1
MMSE: is also linear but requires the noise variance estimation and
channel estimation.
ZF and MMSE with DFB: Direct feedback is used together with
respectively ZF and MMSE to improve their performance. VBLAST is
another name for theses techniques.
MLD: non linear, but provides the maximum likelihood solution. Its
diversity is equal to Nr.
Space Time Block Codes algorithms:
Alamouti: it is a simple algorithm at both transmitter and receiver. It
reaches the optimum capacity but only for a 2x1 system.
Chapter No. 7
MIMO OFDM
7.1 MIMO with OFDM
The previously described SDM algorithms are narrowband single carrier
algorithms. WLAN system are generally broadband and based on orthogonal
frequency division multiplexing (OFDM). OFDM is a multi carrier technique and, within the
standards, the signal time per subcarrier is defined to be TS = 3.2 s. Based on the
observations that indoor rms delay spreads are most likely smaller then 250 ns [Nee00], we
can assume that every subcarrier undergoes flat-fading, since TS is (much) larger than the
rms delay spread [Rap89]. So, in order to combine SDM with OFDM, a SDM algorithm can
be performed per subcarrier.Thus, suppose the transmitter consists of Nt transmit
antennas, then every Sub-carrier carries Nt data streams. At the Nr-th receive antennas the
subcarrier information is separated by using FFTs. After that the Nr information symbols
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belonging to sub-carrier i are routed to the i-th MIMO decoder where one of the algorithms
of is implemented to recover the transmitted data signals (d1,i,, dN,i), where the first
subscript indicates the transmit antenna and the second one indicates the subcarrier
number. This is shown in Figure 7-1 and Figure 7-2.
Figure 7.1 Multi-antenna joint coding architecture transmitter using OFDM.
7.2 Transmitter and receiver design
The transmitting part consists of a multi-antenna transmitter, which is represented
schematically in Figure 7-1. The binary input data is fed to the encoder. A convolutional
code with the IEEE 802.11a standard rate 1/2, constraint length 7 and generator
polynomials (133,171) is chosen as a forward error correction code. Higher coding rates of
2/3 and 3/4 are obtained by puncturing the rate 1/2 code. In order for the forward error
correction to correct for subcarriers and/or antennas that are in deep fades, the coded data
is interleaved over frequency and space to reduce the number of bit errors in one burst.
The interleaver size is chosen based on the assumption that the channel is quasi-static.
The bits are then mapped on QAM symbols according to the IEEE 802.11a standard. The
output of the decoder is then demultiplexed into Nt blocks of Nc streams, where Nc is the
number of sub-carriers. After pilot insertion, the Inverse Fast Fourier Transform (IFFT) is
performed on each block, resulting in Nt signals in the time domain. Cyclic extension and
windowing are based on the IEEE 802.11a standard.
At the receiver, as depicted in Figure 12-2, perfect synchronization (time and
frequency wise) and channel knowledge is assumed. After the payload is received, the
cyclic extension is removed, the FFT is executed and the pilots are removed. Once the
subcarrier information is retrieved, the desired MIMO processing (i.e. ZF, VBLAST or
MLD) can be performed. The output of the MIMO processing is sliced (i.e.,
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demodulated), deinterleaved and depunctured. Finally, the de-punctured bits are decoded
using a Viterbi decoder.
Figure 7.2 Multi-antenna receiver using SDM with OFDM.
Per-Antenna-Coding (PAC) architecture, as illustrated in Figure 7-3, is also
assessed through link level simulations. V-BLAST (Section 4.5.2) is used at the receiver.
Per Antenna Coding V-BLAST is a variant of V-BLAST based on per antenna coding
architecture . The difference with joint coding architecture is that the coding and the
interleaving at the transmitter are now done per antenna branch. At the receiver, the idea is
first to go through the decoding stage before the Successive Interference Cancellation
(SIC) is executed. In this way Forward Error Correcting coding is performed on the SIC
information. In Figure 7-3 and Figure 7-4 a schematic representation of the transmitter and
the receiver of a system deploying PAC V-BLAST is represented. The MIMO OFDM
transmitter consists of Nt OFDM transmitters among which the incoming bits are spread,
then each branch in parallel performs encoding, interleaving (H), QAM mapping, Nc-point
Inverse Fast Fourier Transformation (IFFT), and adds the cyclic extension before the final
TX signal is upconverted to RF and sent.
Figure 7.3 PAC MIMO OFDM transmitter scheme.
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At the Nr receivers, the subcarrier information is separated by performing the Ncpoint Fast
Fourier Transformation (FFT). Then, in general, the symbols mapped onto subcarrier i are
routed to the i-th MIMO detector to recover the M transmitted data signals per subcarrier
(see Figure 7-4). Finally, demapping, deinterleaving (H-1) and decoding are performed per
receiver branch and the resulting data are combined to obtain the binary output data.
Figure 7.4 The PAC V-BLAST Detection and Decoding Block.
7.3 Performance evaluation: noise-limited scenarios
The goal of this section is to present the PER (64 bytes) versus SNR performance of the
MIMO algorithms presented. Although more results are presented in Chapter 6, results for
some test case are presented here to provide a more practical understanding of the
performance of the different MIMO algorithms. The channel used for the simulations is
described in [B4-D2.2] and exhibits a RMS delay of 50ns. The algorithms deployed at the
receiver were ZF, MMSE and MLD.
Figure 12-5 shows the PER (64 byte packet) versus SNR for a 2x2 system using coding
rate 1/2 and QPSK modulation. MLD clearly achieves the best performance followed by
MMSE and ZF. The spectral efficiency achieved by a 2x2 system with 1/2 rate
convolutional code and QPSK modulation is 1.2 bps/Hz. This translates into a data rate of
24Mbps.
In Figure 7.5 coding rate 0.75 and 16QAM modulation are used at the transmitter. The
spectral efficiency achieved by a 2x2 system with 3/4 rate convolutional code and 16QAM
modulation is 3.6 bps/Hz. This translates into a data rate of 72 Mbps.
All curves shift to the right due to the higher data rate that is now transmitted.
MLD still shows the best performance. MMSE loses the advantage over ZF
observed for lower constellations.The performance degradation going from QPSK to
16QAM for MLD and ZF is around 11-12 dB at PER=10-2, while for MMSE is almost 15dB.
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Figure 7.5
PER versus SNR performance of a 2x2 system for different receive algorithms. At the
transmitter QPSK modulation and a convolutional coding rate of 1/2 are used. The RMS
delay spread of the channel is 50ns.
Figure 7.6
PER versus SNR performance of a 2x2 system for different receive algorithms. At the
transmitter 16QAM modulation and a convolutional coding rate of 3/4 are used.The RMS
delay spread of the channel is 50ns.
Figure 7-7 and Figure 7-8 use the same coding rate and modulations as Figure 7-5 and
Figure 7-6, respectively, only then for a 4x4 system. The MLD curve is steeper than in the
previous figures, which is due to the fact that the diversity order, which is proportional to the
number of receive antennas, is now higher. As expected, the slope of the ZF curve does
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not change since the diversity order, which is given by Nt-Nr+1 is still equal to 1 for the 4x4
case. Again for higher constellation MMSE degrades more than MLD and ZF.
The spectral efficiency achieved by a 4x4 system with coding rate equal 1/2 using QPSK is
2.4 bps/Hz, which translates in a data rate equal to 48Mbps. The spectral efficiency
achieved by a 4x4 system with 3/4 rate convolutional code using 16 QAM is 7.2 bps/Hz.
This translates into a data rate of 144 Mbps.
Figure 7.7
PER versus SNR performance of a 4x4 system for different receive algorithms.At the
transmitter QPSK modulation and convolutional coding rate of 1/2 are used. The RMS
delay spread of the channel is 50ns.
Figure 7.8
PER versus SNR performance of a 2x2 system for different receive algorithms.
At the transmitter 16QAM modulation and convolutional coding rate of 3/4 are used. The
RMS delay spread of the channel is 50ns. algorithm, especially for high number of
antennas, followed by MMSE and ZF.
7.4 Performance evaluation: interference-limited scenarios
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The goal of this section is to asses the performance of MIMO based WLAN system in co-
channel interference limited environment. We regard the case where no attempt is made to
cancel interference. First the co-channel interference model is described. Subsequently,
the results from simulation with this model are presented.
Co-channel Interference model
Figure 7.9 Co-Channel Interference Model
MIMO system model in presence of two sources of co-channel interference.
A source of interference has been implemented in MATLAB in order to test the robustness
of a MIMO system in an interference-limited scenario. The interferers are OFDM systems
transmitting in the same bandwidth of the desired user. The number of antennas used by
the interferer can be selected to be single or multiple.
Unless it is specified, the term interference will always be used for co-channel
interference. The interference signal is implemented as a random sequence of modulated
symbols constantly overlapping the desired user signal and with power equal to one. The
time version of the interference signal is convolved with a channel created in a similar way
as the one of the desired user. To come to the right average signal-tointerference ratio
(SIR), the signals from each interference source are multiplied by the square root of a
factor beta defined as:
Where, Nt is the number of transmit antennas of the desired user, Nint is the number of
transmit antennas of the interferer, numint is the number of interferers and SIR is the signal
to interference ratio. With int N num interference sources, the interference power is
int int N num . Whereas the power of the desired user is Nt. Thus we get the following ratio:
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dependent on the number of transit antennas.
Finally, interference, desired signal and White Gaussian noise are added together. In the
simulation presented in next section, the interference source is constantly present during
the transmission of the desired user (i.e. it has the same length D of the desired user
packet and no temporal shift). Moreover the channel of the desired user is supposed ideally
known and for all simulations and its RMS delay spread used is 50ns. When co-channel
interference is considered, the signal model is described by:
7.5 Synchronous versus asynchronous model when the channel is
ideally known
In the simulations that will follow, it is assumed that the interference source is
constantly present during the transmission of the desired user. We will refer to this sort of
interference as synchronous interference. When a uniformly distributed shift in the interval
[-D, D] is created to produce a random temporal overlapping between the desired user and
the interference, the average interference energy is half of the one experienced in the
synchronous interference scenario. This result comes from the fact that the average
overlapping time is D/2. Now lets assume that the channel is ideally known. It can be
shown by means of simulations that the synchronous model and the asynchronous model
produce the same BER versus SIR curves if the power of the asynchronous interference is
doubled. It can be concluded that when the channel is considered perfectly known, the
asynchronous case is a special case of the synchronous one. It is possible by a proper
scaling together the BER performance for the asynchronous case from the synchronous
one.
For this reason in next sections, where channel is assumed perfectly known, the
synchronous interference model has been used for the link level simulations.
Noise to model co-channel interference
When the number of co-channel interferers is high (>>1), the total interference
signal can be more easily modeled as White Gaussian noise. The result of this is a better
BER versus SIR performance, when coding is used. This is shown in Figure.
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Figure 7.10
BER versus SNR performance for SIR=15dB, with 1 OFDM co-channel interference
source, 10 OFDM sources, 1 noise modeled co-channel interference.
In general it is not expected that a high number of co-channel interference sources will be
observed. However, it appears that an interference source exploiting multiple transmitters is
less harmful that one exploiting a single antenna.
7.6 Impact of Co-Channel Interference on MIMO and SISO systems
In this section we analyze the performance of MIMO transceivers in an
interference-limited scenario. Our first goal is to compare the performance of a MIMO and
SISO system in the same interference scenario. Figure 7-11 shows the performance, in
terms of BER versus SNR, of a 1x1 system versus a 2x2 system with different level of co-
channel interference (SIR).
QPSK modulation with no coding is used. There are two sources of interference, both
having a single antenna. It is clear from Figure 12-11 that for high level of interference (SIR
> 10dB) the performance of both SISO and MIMO system is the same. When the SIR =
20dB, the spatial diversity exploited by the MIMO receiver produce a better BER
performance. It is worthy to notice that the spectral efficiency achieved by MIMO is double
of the one achieved by the SISO system. Thus we can conclude that a SISO and a MIMO
system using the same data rate per antenna offer the same performance at low SIR
values.
Figure 7.11 BER vs SNR for a 1x1 and a 2x2 system using QPSK no coding.
For the results shown in Figure 7-12 and Figure 7-13, coding and modulation are chosen in
such a way that the spectral efficiency is the same in both systems. Figure 7-12 shows the
comparison in BER versus SIR and SNR, of a 2x2 system using BPSK modulation and
coding rate and a 1x1 system using QPSK modulation and coding rate . In both cases
the spectral efficiency is equal to 1,2 bps/Hz. We can conclude that for a given spectral
efficiency, a MIMO system is more robust to interference than a SISO system. This is due
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to the fact that MIMO uses lower modulation schemes (per antenna) than SISO to achieve
the same data rate. At system level more packets will be correctly delivered by a MIMO
system increasing the total throughput. Figure 5-13 shows the same as 2x2 BPSK vs 1x1
QPSK, R=1/2. Both systems achieve the same spectral efficiency, equal to 2.4bps/Hz.
Figure7.12
2x2 BPSK vs 1x1 QPSK, R=0.5. Both systems achieve the same spectral efficiency,
equal to 2.4bps/Hz.
Figure 7.13
4x4 BPSK vs 1x1 16QAM, R=1/2. Both systems achieve the same spectral efficiency,
equal to 2.4bps/Hz.
7.7 Robustness of MIMO to co-channel interference
Figure 7.15 and 7.16 show the BER performance versus SIR of a 2x2 system using PAC V-
BLAST. BPSK modulation and a half rate code have been used. The difference between
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the two figures is the RMS delay spread: which is respectively 50ns and 250ns. For both
simulations two interference sources has been considered: a single antenna transmitter
(indicated in the legend of the figures below as SIMO) and a multiple antenna transmitter
one (indicated in the legend of figures below as MIMO). The figures depicted below show
the BER performance versus the SIR for the given SNR. The SNR value is chosen to get a
BER around 10-4 or lower. As expected, all the curves tend to reach, for high SIR values
(no interference), the BER value at the chosen SNR, which is identified in each figure by
the asymptote. Note that in PAC V-BLAST the ideal knowledge of beta is assumed. So in
order to minimize the Mean Square Error (over D), the processing at the receiver is equal
to:
where equals Nt/SNR and is as defined as in (4), where H is the NrxNt channel matrix.
For low delay spread, there is no difference in performance when the reception of a 2x2
system is corrupted by a single transmit antenna interference or by a multiple transmit
antenna interference.
At a delay spread of 250 ns the performance in presence of a multiple antenna interferer is
slightly better than in the case of a single antenna interferer. This is due to the fact that a
higher number of interferers make the spectrum of the overall interferer signal flatter over
frequency. For high delay spreads and when coding is used, this kind of spectrum has a
lighter impact, on average, on each subcarrier of the received signal. This is more visible in
Figure 7-17 for a 4x4 system.
Figure7.15
BER versus SIR for a 2x2 system. The RMS delay spread is 50 ns and the Eb/No is fixed
at 5 dB. The source of interference is one terminal with either one or two transmit antennas.
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Figure 7.16
BER versus SIR for a 2x2 system. The RMS delay spread is 250 ns and the Eb/No is fixed
at 5 dB. The source of interference is one terminal with either one or two transmit antennas.
Figure 7.17 and Figure 7.18 depict the same results as Figure 7-15 and Figure 7-16, only
now for the 4x4 case. Here the source of interference is either a single transmit antenna
system or a four transmit antenna system. For low delay spread there is no difference
between the performances in presence of different sources of interference.
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Figure 7.17
BER versus SIR for a 4x4 system. The RMS delay spread is 50 ns and the Eb/No is fixed
at 0 dB. The source of interference is one terminal with either one or four transmit
antennas.
Figure7.18
BER versus SIR for a 4x4 system. The RMS delay spread is 250 ns and the Eb/No is fixed
at 1 dB. The source of interference is one terminal with either one or two transmit antennas.
For high delay spread, as previously seen, the system is more robust when a
multiple transmit antenna interferer is present. This is due to the flatter spectrum of the interference signal. From the
figures above, we can conclude that at a SIR of 20 dB, the degradation on performance due to the interference is
negligible; at a SIR of about 10 dB, the BER performance looses 1 decade and for values of SIR lower than 10 dB the
BER performance of PAC V- BLAST rapidly decrease (circa 1 decade per 2 dB for 250 ns delay spreads and 1decade
for 4 dB for 50 ns delay spread). It is important to notice that these results were obtained for the worse case scenario in
which the interference is constantly present.