CHAPTER 1

INTRODUCTION
1.1 Purpose

Efficient use of radio spectrum includes placing modulated carriers as close as

possible without causing Inter-Carrier Interference (ICI). Optimally, the bandwidth of

each carrier would be adjacent to its neighbors, so there would be no wasted spectrum. In

practice, a guard band must be placed between each carrier bandwidth to provide a space

where a filter can attenuate an adjacent carrier’s signal. These guard bands are wasted

bandwidth. In order to transmit high data rates, short symbol periods must be used. The

symbol period is the inverse of the baseband data rate (T = 1/R), so as R increases, T

must decrease. In a multi-path environment, a shorter symbol period leads to a greater

chance for Inter-Symbol Interference (ISI). This occurs when a delayed version of

symbol ‘n’ arrives during the processing period of symbol ‘n+1’.

Orthogonal Frequency Division Multiplexing (OFDM) addresses both of these

problems. OFDM provides a technique allowing the bandwidths of modulated carriers to

overlap without interference (no ICI). It also provides a high date rate with a long

symbol duration, thus helping to eliminate ISI. OFDM may therefore be considered as a

candidate modulation technique in a broadband, multi-path environment. One of the main

reasons to use OFDM is to increase the robustness against frequency selective fading or

narrowband interference. In a single carrier system, a single fade or interferer can cause

the entire link to fail, but in a multicarrier system, only a small percentage of the sub-

carriers will be affected. Error correction coding can then be used to correct for the few

1
erroneous sub-carriers. OFDM along with forward error correction is called Coded

Orthogonal Frequency Division Multiplexing (COFDM)[8].

1.2 COFDM Overview

OFDM is a modulation technique where multiple low data rate carriers are

combined by a transmitter to form a composite high data rate transmission. Digital signal

processing makes OFDM possible. To implement the multiple carrier scheme using a

bank of parallel modulators would not be very efficient in analog hardware. However, in

the digital domain, multi-carrier modulation can be done efficiently with currently

available DSP hardware and software. Not only can it be done, but it can also be made

very flexible and programmable. This allows OFDM to make maximum use of available

bandwidth and to be able to adapt to changing system requirements.

Each carrier in an OFDM system is a sinusoid with a frequency that is an integer

multiple of a base or fundamental sinusoid frequency. Therefore, each carrier is like a

Fourier series component of the composite signal. In fact, it will be shown later that an

OFDM signal is created in the frequency domain, and then transformed into the time

domain via the Discrete Fourier Transform (DFT).

Two periodic signals are orthogonal when the integral of their product, over one

period, is equal to zero. Irregular Turbo codes have been known to attain near Shannon

limit performances in AWGN. Coded Orthogonal Frequency Division Multiplexing

(COFDM), originally developed for digital audio broadcasting, has been shown to be a

very good scheme to combat fading and non-Gaussian noise. It is quite natural then to

look at the combination of COFDM with irregular turbo codes as a system architecture

for mobile radio or indoor environments.

2
 CHAPTER 2

LITERATURE SURVEY

2.1 OFDM

2.1.1OFDM Overview

OFDM is a modulation technique where multiple low data rate carriers are

combined by a transmitter to form a composite high data rate transmission. Digital signal

processing makes OFDM [10] possible. To implement the multiple carrier scheme using

a bank of parallel modulators would not be very efficient in analog hardware. However,

in the digital domain, multi-carrier modulation can be done efficiently with currently

available DSP hardware and software. Not only can it be done, but it can also be made

very flexible and programmable. This allows OFDM to make maximum use of available

bandwidth and to be able to adapt to changing system requirements.

Each carrier in an OFDM system is a sinusoid with a frequency that is an integer

multiple of a base or fundamental sinusoid frequency. Therefore, each carrier is like a

Fourier series component of the composite signal. In fact, it will be shown later that an

OFDM signal is created in the frequency domain, and then transformed into the time

domain via the Discrete Fourier Transform (DFT). Two periodic signals are orthogonal

when the integral of their product, over one period, is equal to zero. This is true of

certain sinusoids as illustrated in equation (1)

3
 Continuous Time :
T
0 cos(2nf 0 t )  cos(2mf0 t )dt  0 (n  m)

(1)
Discrete Time :
N -1
 2kn   2km 
 cos N 
  cos
 N 
  0 (n  m)
k 0

The carriers of an OFDM system are sinusoids that meet this requirement because

each one is a multiple of a fundamental frequency. Each one has an integer number

of cycles in the fundamental period.

2.1.2 OFDM OPERATION

When the DFT (Discrete Fourier Transform) of a time signal is taken, the

frequency domain results are a function of the time sampling period and the number of

samples as shown in Figure . The fundamental frequency of the DFT is equal to 1/NT

(1/total sample time). Each frequency represented in the DFT is an integer multiple of

the fundamental frequency. The maximum frequency that can be represented by a time

signal sampled at rate 1/T is fmax = 1/2T as given by the Nyquist sampling theorem. This

frequency is located in the center of the DFT points. All frequencies beyond that point

are images of the representative frequencies. The maximum frequency bin of the DFT is

equal to the sampling frequency (1/T) minus one fundamental (1/NT).

The IDFT (Inverse Discrete Fourier Transform) performs the opposite operation

to the DFT. It takes a signal defined by frequency components and converts them to a

time signal. The parameter mapping is the same as for the DFT. The time duration of the

IDFT time signal is equal to the number of DFT bins (N) times the sampling period (T).

It is perfectly valid to generate a signal in the frequency domain, and convert it to a time

domain equivalent for practical use. This is how modulation is applied in OFDM. In

4
practice the Fast Fourier Transform (FFT) and IFFT are used in place of the DFT and

IDFT, so all further references will be to FFT and IFFT.

5
s(t)

N
(number of samples)

1 2 3 ........

NT
T (sample period) (total time used for the DFT is the product t
of the sample period times the number of samples)

DF IDF
T T

| S(f) |
........ ........

1/2T
(Nyquist bin)

0 1/NT 2/NT 3/NT …………

……….. (N-1)/NT f
DFT bins representing discrete
frequency components of f(t).
(N/NT = 1/T =
sampling frequency)

Figure 2.1: Parameter Mapping from Time to Frequency for the DFT

6
 Definition of Carriers

The maximum number of carriers used by OFDM is limited by the size of the IFFT.

This is determined as follows in equation (2)

IFFTsize
N carriers  2 (real - valued time signal)
2 (2)
N carriers  IFFTsize  1 (complex - valued time signal)

In order to generate a real-valued time signal, OFDM (frequency) carriers must be

defined in complex conjugate pairs, which are symmetric about the Nyquist frequency

(fmax). This puts the number of potential carriers equal to the IFFT size/2. The Nyquist

frequency is the symmetry point, so it cannot be part of a complex conjugate pair. The

DC component also has no complex conjugate. These two points cannot be used as

carriers so they are subtracted from the total available.

If the carriers are not defined in conjugate pairs, then the IFFT will result in a time

domain signal that has imaginary components. This must be a viable option as there are

OFDM systems defined with carrier counts that exceed the limit for real-valued time

signals given in equation (2). This design must result in a complex time waveform.

Further processing would require some sort of quadrature technique (use of parallel sine

and cosine processing paths). In this report, only real-value time signals will be treated,

but in order to obtain maximum bandwidth efficiency from OFDM, the complex time

signal may be preferred (possibly an analogous situation to QPSK vs. BPSK).Equation

(2), for the complex time waveform, has all IFFT bins available as carriers except the DC

bin.

Both IFFT size and assignment (selection) of carriers can be dynamic. The

transmitter and receiver just have to use the same parameters. This is one of the

7
advantages of OFDM. Its bandwidth usage (and bit rate) can be varied according to

varying user requirements. A simple control message from a base station can change a

mobile unit’s IFFT size and carrier selection.

2.2 COFDM

2.2.1 COFDM overview

Coded Orthogonal Frequency Division Multiplexing (COFDM) [9] has been

specified for digital broadcasting systems for both audio -- Digital Audio Broadcasting

(DAB) and (terrestrial) television -- Digital Video Broadcasting (DVB-T). COFDM is

particularly well matched to these applications, since it is very tolerant of the effects of

multipath (provided a suitable guard interval is used). Indeed, it is not limited to 'natural'

multipath as it can also be used in so-called Single-Frequency Networks (SFNs) in which

all transmitters radiate the same signal on the same frequency. A receiver may thus

receive signals from several transmitters, normally with different delays and thus forming

a kind of 'unnatural' additional multipath. Provided the range of delays of the multipath

(natural or 'unnatural') does not exceed the designed tolerance of the system (slightly

greater than the guard interval) all the received-signal components contribute usefully.

Multipath (natural and unnatural) can alternatively be viewed in the frequency

domain as a frequency selective channel response. Another frequency-dependent effect

for which COFDM offers real benefit is the presence of isolated narrow-band interfering

signals within the signal bandwidth. Note that conventional analogue television signals

(NTSC/PAL/ SECAM) essentially behave like narrow-band interferers to COFDM.

COFDM copes with both these frequency-dependent effects as a result of the use

of forward error coding. However, rather more is involved than simply adding coding --

8
 the 'C' -- to an uncoded OFDM system. The coding and decoding is integrated in a way

which is specially tailored to frequency-dependent channels and brings much better

performance than might be thought based on a casual inspection.

COFDM is a modulation scheme, which is especially tailored to work well with

selective channels and isolated CW (or analogue TV) interferers. The forward error-

correction coding -- the 'C' in COFDM -- is the key ingredient. However, the desired

results are only achieved when the coding is closely integrated with the OFDM system.

2.2.2 COFDM transmitter

cos(2πfct)

M
A Real
P S P
Bit P IDFT
Input I
/ /
N P S Img
G

sin(2πfct)
Generate
Output
d0~dN-1 BPF

Figure 2.2: COFDM transmitter

COFDM modulator:

After the symbol mapping is carried out, the frequency interleaving will re-order

the symbols and then the complex numbers that represent the symbols to be transmitted

on each of the sub-carriers will be sent to a serial-to-parallel converter and "placed" on

each of the sub-carriers.

9
 Then IFFT takes a signal defined by frequency components and converts them to

a time signal. After the IFFT has been calculated, the output complex numbers are

parallel to serial converted, and following this the cyclic prefix (or guard period) is

inserted and then transmitted through the channel.

2.2.3 COFDM receiver

LPF
F
R
O A S P
N cos(2πfct)
Input / DFT
T D / /
E
D
π/2
P S
LPF

M
A
P
Bit Output P
I
N
G

Figure 2.3: COFDM receiver

After the signals are received at the antenna, the signals are down converted from

RF to generate the real (I) and imaginary (Q) streams, lowpass filtered (LPF) and

digitized in the analogue to digital converters (ADC). Following the ADC, the cyclic

prefix is stripped off and the remaining sampled values are serial to parallel converted

and once there is a full block of samples the FFT is calculated. After the FFT (the FFT is

10
the OFDM demodulator), the received symbols are parallel to serial converted and

interleaving is reversed and is sent to error correction decoder.

2.3 FORWARD ERROR CORRECTION (FEC)

Forward error correction (FEC) relies on the controlled use of redundancy in the

transmitted code word for both the detection and correction of errors incurred during the

course of transmission over a noisy channel. Irrespective of whether the decoding of the

received code is successful, no further processing is performed at the receiver.

Accordingly, channel coding techniques suitable for FEC require only a one-way link

between the transmitter and receiver.

The FEC section of an OFDM system can be addressed by either a block-based

coding scheme (Reed-Solomon) or a convolutional coding scheme (Viterbi, Turbo).

OFDM systems using FEC techniques are also referred to as coded-OFDM (COFDM)

transmitters. FEC enables the receiver to correct errors automatically without requesting

re-transmission. FEC is based on adding redundant parity information to the data being

transmitted, but in this case the receiver not only detects that an error has occurred, but

calculates what information the transmitter is most likely to have transmitted. Majority of

COFDM systems also use an interleaver (block or convolutional), which minimizes the

burst errors within the data channel. The interleaved data is then passed through a serial-

to-parallel converter, which maps the symbols onto an IQ constellation specific to the

modulation scheme.

11
2.3.1 Irregular turbo codes

Irregularity

Normally in any encoding operation, the input bits are directly fed to the encoder.

These are called as regular codes, since each input bit is mapped as a unique input bit of

encoder. Block codes, cyclic codes, RS codes, Turbo codes are examples of regular

codes. The process of repeating some of systematic bits before encoding is called as

called as irregularity.

The irregularity can be introduced in the turbo code. A turbo code using this

concept is called as irregular turbo code. Irregular turbo codes were found by B.J. Frey

and D.C.Mackay in 1999. These codes are actually derived from original turbo code. A

classical turbo code has two recursive systematic convolutional (RSC) constituents

separated by an interleaver. The encoder has 3 branches. Systematic bits, First RSC

(upper) constituent and second RSC (lower) constituent. The interleaver shuffles the data

i.e. changes the data bit order and then feds to RSC. Each systematic bit is copied exactly

once. In classical turbo the degree of all information bits is 2. So in regular turbo codes

each systematic bit participates exactly in 2 trellis sections. If a change is made such that

some systematic bits are repeated before feeding it to RSC, Irregular turbo code is

obtained. For example each of 10 percent of the systematic bits may be mapped to eight

inputs of convolutional encoder instead of a single bit.

12
2.3.2 Irregular turbo code structure

The structure of irregular turbo code is slightly different from normal turbo code.

Irregular turbo codes use a special design of interleaver that maps some systematic bits to

multiple input bits of the convolutional encoder. The error correcting performance of

turbo codes over a noisy channel can be improved by using irregular turbo codes. It is

just 0.213dB away from Shannon’s theoretical limit.

By slight modifications, the original turbo can be made irregular [1, 2]. Figure 4

shows how a turbo code can be made as irregular. Figure 2.4a shows the set of systematic

bits (middle rows of discs) being fed directly into one convolutional code (the chain at

the bottom) since the order of the systematic bits is irrelevant, an interleaver may be

introduced before upper convolutional code as shown in figure 2.4b. Figure 2.4c is same

as figure 4b except that two interleavers and convolutional code are shown side by side.

13
 Interleaver

(a) (b)
Interleaver Interleaver

Interleaver Interleaver (c)

Interleaver d

Interleaver e

Fig2.4: Irregular Turbo code

14
 For long codes, the values of initial state and final state of the convolutional code

do not significantly influence the performance. So turbo code may be viewed as a code

that copies the systematic bits, interleaves both sets of these bits, and then feeds them into

convolutional code.

Some systematic bits may be tied together, in effect causing some systematic bits

to be replicated more than once. This is as shown in fig 2.4e. Even though, some bits are

repeated, some bits are punctured to make over all rate fixed.

A more general form of irregular turbo code can be obtained by cascade of array

of repetition codes, an interleaver and a turbo code. This is illustrated as in figure 2.5

Repetition Irregular turbo

Interleaver Turbo code
codes code

Figure 2.5: General form of irregular turbo code

2.3.3 Parameters of irregular turbo code

The degree profile (fd) and maximum degree (D) are important parameters of

irregular code. fd is the fraction of codeword bits that have degree d , f d  [0,1],d 
{1,2,3….D}, D is the maximum degree. Each codeword bit with degree d is repeated d

times before being fed into the interleaver. The average code word degree is given as in

equ (3)

D
d =  d.fd (3)
d1

The minimum degree is 2 is for irregular turbo code. The sum of fraction of all degrees is

1. This is as shown in equ (4)

D
 f 1 (4)
d
i2

15
The over all rate R of an irregular turbo code is related to the rate R  of the

convolutional code and the average d as in equ (5)

d (1  R  )  1  R (5)

So, if average degree is increased, the rate of the convolutional code must also be

increased to keep the over all rate constant. This can be done by puncturing the

convolutional code or by designing a new, higher rate convolutional code.

2.3.4 Decoding of irregular turbo codes

The decoding scheme used for irregular turbo code is slightly different from

original turbo decoding. The algorithm is as follows

1) After receiving channel output, the channel output log likelihood ratios for the N

codeword bits ( L01, L02 ,......, L0N ) is computed.

2) If the codeword bit i has degree d, then log likelihood ratios are assigned as in equ

(6)

L  L0 , L  L0 ..... L  L0 (6)
i,1 i i,2 i i, d i

the likely hood ratios are interleaved and fed to MAP algorithm. The Map algorithm

assumes that inputs are priori log probability ratios and then computes a set of

posteriori log-probability ratios Li,1, Li,2 ,....Li, d . For regular code turbo code, there

are just two a posteriori probability ratios L and Li, d , for each they correspond to
i,1

extrinsic information produced by each constituent convolutional code

3) The current estimate of the log-probability ratio for bit i given the channel output

is given as in equ (7)

d
L̂  L0   (L  L ) (7)
i i i, k i, k
k 1

16
 4) To compute the inputs to the MAP algorithm needed for next iteration,

subtraction off corresponding outputs from the MAP algorithm produced by the

previous iteration as in equ (8)

L  L̂  L (8)
i, k i i, k

This completes first iteration. The steps from 1 to 4 are repeated for required

iterations and taking repetitions into account to combine the outputs to form estimates of

the log-probability ratios of the codeword.

Irregular turbo decoder operates on noisy versions of the systematic bits and the two

sets of parity check bits in two-decoding stages to produce an estimate of the original

message bits. Each of the two decoding stages uses a BCJR algorithm to solve a

maximum a posteriori probability (MAP) detection problem.

The BCJR algorithm is a soft input-soft output decoding algorithm with two

recursions, one forward and the other backward, both of which involve soft decisions.

The irregular turbo code decoders often work iteratively (loop wise) by sharing the a

priori information obtained from the log-likelihood ratio (a posteriori information) of the

previously cascaded decoder. This sharing of information is made possible through the

interleaving involved in the encoder that creates two weakly correlated parity streams.

For irregular turbo codes, the Soft Output Viterbi Algorithm (SOVA), and the Log-

MAP decoding algorithm can be used as they produce soft-bit estimates. The Log-MAP

decoding scheme is the modified version of the MAP decoding scheme and is

computationally less complex than the original MAP decoding algorithm. The MAP

decoding algorithm was first introduced (in 1974) by Bahl, Cocke, Jelinek and Raviv and

is also known as the BCJR algorithm. However, due to the push for strikingly low bit

17
error rates, the MAP or the Log-MAP has been most commonly used in irregular turbo

codes since they are based on the optimal decoding rule [1]. In contrast, the SOVA [3] is

an approximation to the MAP sequence decoder and will have a slightly worse bit error

performance. Though SOVA suffers from performance degradation as opposed to the

Log-MAP decoding rules, it has much reduced complexity.

MAP Algorithm

To first understand the decoding of turbo codes, a preliminary understanding of

the MAP (BCJR) algorithm is necessary. In the original paper, the idea was set out to

estimate the a posteriori probabilities of the states and transitions of a Markov sequence

transmitted through a discrete memoryless channel. This work resulted in an algorithm

that minimizes symbol error rates while trying to decode block and trellis codes.

The aim of the MAP algorithm is to minimize the symbol error rate for the

decoding of trellis and block codes. Therefore, after receiving the information through the

channel, the job of the decoder is to determine the most likely input bits

(original/uncoded information sequence), based on the received symbols. Since the input

is over the binary alphabet, it is conventional to form a log-likelihood ratio (LLR) and

base the bit estimates on comparisons based on magnitude of the likelihood ratio to a

threshold. The log-likelihood ratio for the input symbol indexed at time t is defined as

The decoder produces estimates of the information bits based on the values of the log-

likelihood ratio. The magnitude of the log-likelihood ratio is defined as the soft output or

soft value which can be passed after processing to the other decoder as an apriori

18
information. Also, the sign of the LLR determines the hard estimate of the original

information sequence. The estimator obeys the following rule:

In order to perform the decoding when the information is received through the

channel, the log-likelihood ratio must be computed.

2.3.5 Puncturing

Puncturing is used to delete one or more coded parity bits from a codeword in a

code. This well known technique helps to obtain high-rate codes without modifying the

structure of the encoder and the decoder circuit for an existing encoder/decoder circuit.

Although punctured codes makes a system flexible by allowing the change in the code

rate they usually suffer from performance degradation as compared to the unpunctured

codes.

2.3.6 Interleaving

The decoding algorithm performs poorly when it is presented with bit errors that

are all bunched together in the stream. Because the sub-carriers are subject to flat fading

bit errors usually do occur in groups when a sub-carrier is in a deep fade. To protect

against this, COFDM uses time interleaving and frequency interleaving.

Interleaver Design

An interleaver is a device that mixes up the symbols from several codewords so

that the symbols from any given codeword give a random effect. When the deinterleaver

19
reconstructs the codeword by arranging the received sequence to its original order, error

bursts introduced by the channel are broken up and spread across several code words.

Interleavers were used practically in both serially concatenated codes and in multipath

fading channels to enhance the overall error correcting capability of the coding scheme.

They were used differently for the first time in a parallel concatenated code and proved to

reduce the number of codewords with small distances in the code distance spectrum (i.e.

generates less code words with minimum hamming distance).

(a) Golden Relative Prime Interleaver

The golden relative prime interleaver computes the scrambler indexes using the

following relation

………….(9)

where, s is an integer starting index, p is an integer index increment, and N is the

interleaver length. The values of N and p must be relative primes in order to ensure the

uniqueness of each index element. The starting index s is usually set to 0, but other

integer values can also be selected. The relative prime increment p is chosen close to one

of the non-integer values of c, which is computed as shown below

…………………………(10)

In the expression of c, g is the golden section value, m is any integer greater than zero, r

is the index spacing, and j is any integer modulo r. The preferred values of j, m and r are

0, 1 and 1 respectively.

(b) Golden Interleaver

The Golden interleaver does not depend on the usage of relative primes and

integer modulo arithmetic but is rather based on the principle of sorting real-valued

20
numbers derived from the golden section value. After computing the value of c as given

by equation (10) the real-valued golden vector e is calculated as follows:

………………….(11)

where, s is any real starting value. After finding the vector e it is possible to find the

index vector z. If the sorted version of the vector e is denoted by a, then the index vector

z and the sorted vector a will be relates as shown below:

………………………………(12)

finally, the golden interleaver indexes are then computed by the following expression:

……………….………..(13)

2.4 WIRELESS CHANNEL MODEL

A communication channel is defined as a physical medium over which

information can be transmitted .The channel may be wired or wireless. Fiber optic,

coaxial cables are wired channels where as ionospheric propagation, free space are

examples of wireless channels. The factors which are responsible for corruption of

signals in channels are attenuation, non-linearity, and multipath propagation. In order to

study the effects of these factors there are some well-known channel models among

which the Gaussian (AWGN) and slow flat independent Rayleigh fading channel are

prominent.

2.4.1 Additive White Gaussian Noise wireless Channel

21
 The Additive White Gaussian Noise (AWGN) channel is one of the most

commonly used channel model and is generally used to model an environment with a

very large number of additive noise sources. Most additive noise sources in modern

electronics are a direct consequence of zero-mean thermal noise, which is caused by

random electron motion within the resistors, wires, and other components. By the Central

Limit Theorem, we can model these additive sources as a Gaussian random process. This

assumption becomes more accurate as the number of noise sources is increased. The

statistical model for the AWGN channel with zero mean is given by its probability

density function in equ (14) along with its variance (  2) (for BPSK modulation) in

equ(15)

2
1 x
p(x)  exp( ) (14)
2  2 2 2

N
0
 2

2RE
b
(15)

number of information bits Eb

Where, R is the coding rate ( R  ) and is the ratio of the
number of transmitted bits N0

bit energy to noise power spectral density. The modulated and the received signal r at

anytime instant t is given in equ (16)

rt  x t  n t (16)

Where xt represents the information sequence and nt is the white noise added by the

channel at time t.

22
2.4.2 Multipath fading wireless channels

In a mobile environment, the surrounding objects such as buildings, other moving

vehicles and trees act as reflectors of radio waves. These obstacles produce reflected and

scattered waves with attenuated amplitude and phase shift. For one transmitted there will

be multiple reflected waves that arrive at the receiving antenna from different directions

with different propagation delays. This type of transmission is called multipath

propagation. Due to the different arrival angles and times, the multipath waves at the

receiver will have different phases. When the receiver antenna at any point in space

collects them, they may combine either in a constructive or a destructive way, depending

on their random phases. The sums of these multipath components form a spatially

varying standing wave field. The mobile unit moving through the multipath field will

receive a signal, which can vary widely in amplitude and phase. When the mobile unit is

stationary, the amplitude variations in the received signal are due to the movement of

surrounding objects in the radio channel. The amplitude fluctuation of the received signal

is called signal fading and it is caused by the time-variant multipath characteristics of the

channel. In general, a multipath fading channel’s characteristics can be modeled using a

tapped delay line model where the tap coefficients, c i’s, as being complex variables
Input Channel
characterizing the low Delay
pass complex Delay of the Delay
envelope represented signal.
signal outputA general

multipath fading channel is shown in Figure 2.6.

C1(t) C2(t) CL-1(t) CL(t)

23 Additive noise

Fig 2.6:Tapped delay line for multipath fading channel

 The various mobile channel parameters are Maximum Excess Delay, Root Mean

square Delay Spread, Coherence Bandwidth, Doppler Spread and Coherence Time.

2.4.3 Fading Effects Due To Multipath

Multipath fading channels exhibit different behaviors. The channel behavior is

mainly characterized by the time delay spread and the Doppler spread. The fading effects

due to these parameters are described as follows:

(1) Flat Fading

Flat fading implies that the signal bandwidth is small as compared to the channel

bandwidth so that the received signal will have varying amplitude according to the gain

fluctuations of the channel. In case of flat fading, the coherence time of the faded symbol

is much smaller than the symbol period of the transmitted signal. For flat fading mobile

radio channels, the phase response is linear and hence, the receiver is able to track the

phase variations, and hence can compensate for it using a phase locked loop technique.

This enables coherent detection of the received signal.

24
(2) Frequency Selective Fading

Frequency selective fading occurs in a situation where the channel’s bandwidth is

smaller than the bandwidth of the transmitted signal. For frequency selective fading the

coherence time is larger than the symbol period. A common rule of thumb to determine

that a channel is frequency selective is that the symbol period should be at least ten times

smaller than the root mean square delay spread. Frequency selective fading occurs due to

the time dispersion of the transmitted symbols within the channel, and hence, induces

intersymbol interference.

(3) Slow Fading

For a slow fading channel, the channel’s impulse response exhibit slower

variations than the transmitted baseband signal. In slow fading channels, the coherence

time of the channel is greater than the symbol period of the transmitted signal. In

frequency domain it implies that the Doppler spread of the channel is much less than the

bandwidth of the baseband signal. In this situation, the channel can be understood to be

static over one or several reciprocal bandwidth intervals.

(4) Fast Fading

In a fast fading channel, the channel impulse response varies rapidly within the

symbol duration. In technical terms this implies that the coherence time of the channel is

smaller than the symbol period of the transmitted signal or, equivalently, the Doppler

spread of the channel is greater than the bandwidth of the transmitted signal.

2.4.4 Rayleigh Fading Wireless Channel Model

25
 A slow flat independent Rayleigh fading channel model is used for the fading

environment. Rayleigh fading is commonly used statistical time varying nature of the

received signal of flat fading signal or the envelope of individual multipath component.

An independent Rayleigh fading process can be modeled as a constant random variable

during each symbol interval. In order to introduce the effect of independent fading

channel, a random Rayleigh distributed number is generated and to this number, AWGN

is added to imitate the complete effect caused by the channel. The modulated and the

received signal r at any time instant t is given as in equ (17)

rt  a x  n ( 17)
t t t

Where, at is a Rayleigh random variable that represents the amplitude variation of the

fading channel, n is the AWGN added by the channel, x is the input to the channel
t t

and r is resulting envelope due to transmission. The statistical model for the envelope
t

of the Rayleigh random variable is given by its probability density function in equ (18)

 a a2
 exp( ), a  0 (18)
p(a)    2
2 2
 0 a 0


It is well known that the envelope of the sum of two quadrature Gaussian noise signals

obeys a Rayleigh distribution. So the fading envelope is given as in equ (19)

at  I2  Q2 (19)

2.4.5 Channel capacity

26
 Channel capacity is the maximum rate at which reliable transmission of

information over the communication channel is possible. According to Shannon, reliable

transmission is possible when there exists a sequence of codes with increasing block

length, for which the error probability tends to zero as the block length increases. This is

called Channel coding theorem.

According to Shannon, if C is the channel capacity and R is the data rate then
RC RC
reliable transmission is possible when. When there exists

a code capable of achieving an arbitrarily low probability of error. When R>C, it is not

possible to have reliable transmission.

Shannon defined the channel capacity of AWGN channel as equ (20)

S
C  Blog (1  ) (20)
2 N

Where B is channel bandwidth in hertz and S/N is the signal-to-noise power ratio.

This formulation is known as Shannon-Hartley law. For Infinite signal-to-noise ratio

(noise less case), channel capacity is infinite for any non-zero bandwidth. The energy per

bit (Eb) is equal to the bit time (T b) multiplied by signal power S. At capacity, the bit rate

(Rb) is equal to the capacity. Thus T b=1/C seconds per bit. These yields, at capacity by

equ (21)

Eb=S Tb=S/C (21)

The total noise power in bandwidth B is given by equ (22)

N=N0B (22)

Where N0 is the single sided noise power spectral density in watts per hertz. The signal-

to-noise can be expresses as equ (23)

27
 S Eb C
 (23)
N N0 B

This allows the Shannon-Hartley law to written as in equ (24)

C  log 2(1  Eb C ) (24)

B N0 B

Solving (24), Eb/ N0 is given as equ (25)

C
Eb B
 ( 2 B  1) (25)
N0 C

(25) establishes performance of ideal system. For the case in which B>>C, E b/ N0 is

given in equ (22) and (23)

C
C C
2 B  exp( ln(2))  1  ln(2) (26)
B B

Where the approximation exp(x)  1  x , for x  1 is used to get equ (26)

Substituting (24) in (23) for Eb/ N0 is as in equ (27)

Eb
 ln(2)  1.6 dB, B  C (27)
N0

Thus for an ideal system, in which Rb =C, Eb/ N0 approaches the limiting value of

-1.6 dB as the bandwidth grows without bound. Hence -1.6 dB is the minimum required

S/N per bit to achieve small probability of error. The minimum S/N per bit (-1.6 dB) is

called Shannon limit for an AWGN channel. Any coding scheme designed should operate

close to this limit for efficiently using channel capacity with minimum probability of

28
error. Since wireless channel is prune to burst errors, it is necessary to adopt efficient

coding schemes like turbo code and irregular turbo codes.

CHAPTER 3

29
 COFDM OPERATION
3.1 Overall structure of the transmission system

The general architecture of the system is shown in Figure 3.1. In this scheme,

irregular turbo codes are used for channel encoding and COFDM is the modulation

method.

IFFT
Binary Channel Time Serial Signal mapping Frequency +
data coding Interleaving to (Baseband Interleaving Guard
parallel modulation) interval

Channel

Output Channel Time De- Parallel Frequency

Decoding Interleaving To Demodulation De- FFT
Serial Interleaving

Figure 3.1: General Structure Of COFDM System.

3.2 Forward Error Correction

Forward error correction (FEC) relies on the controlled use of redundancy in the

transmitted code word for both the detection and correction of errors incurred during the

course of transmission over a noisy channel. Irrespective of whether the decoding of the

received code is successful, no further processing is performed at the receiver.

30
Accordingly, channel coding techniques suitable for FEC require only a one-way link

between the transmitter and receiver. Here irregular turbo codes have been used for FEC.

Irregular Turbo Codes

Irregular Turbo Encoder

In its most basic form, the encoder of an irregular turbo code consists of two

systematic encoders joined together by means of two irregular interleavers as illustrated

in figure 3.2.

Figure 3.2: Irregular turbo encoder

Typically, but not necessarily, the same code is used for both constituent encoders in

Figure 3.2. The constituent codes recommended for irregular turbo codes are short

constraint-length recursive systematic convolutional (RSC) codes. The reason for making

the convolutional codes recursive is to make the internal state of the shift register depend

on past outputs. This affects the behavior of the error patterns, with the result that a better

performance of the overall coding strategy is attained.

31
 Irregular turbo codes use a special design of interleaver that maps some systematic

bits to multiple input bits of the convolutional encoder. For example, each of 10 percent

of the systematic bits may be mapped to eight inputs of the convolutional encoder instead

of a single one. As shown in Figure 3.2, irregular interleavers are used in both

convolutional encoding paths to generate the parity-check bits z1 and z2 in response to

the message bits x. The error correcting performance of turbo codes over a noisy channel

can be improved by using irregular turbo codes. It is just 0.213dB away from Shannon’s

theoretical limit.

Irregular Turbo Decoding

Irregular turbo decoder operates on noisy versions of the systematic bits and the two

sets of parity check bits in two-decoding stages to produce an estimate of the original

message bits. Each of the two decoding stages uses a BCJR algorithm to solve a

maximum a posteriori probability (MAP) detection problem.

The BCJR algorithm is a soft input-soft output decoding algorithm with two

recursions, one forward and the other backward, both of which involve soft decisions.

The irregular turbo code decoders often work iteratively (loop wise) by sharing the a

priori information obtained from the log-likelihood ratio (a posteriori information) of the

previously cascaded decoder. This sharing of information is made possible through the

interleaving involved in the encoder that creates two weakly correlated parity streams.

For irregular turbo codes, the Soft Output Viterbi Algorithm (SOVA), and the Log-

MAP decoding algorithm can be used as they produce soft-bit estimates. The Log-MAP

decoding scheme is the modified version of the MAP decoding scheme and is

computationally less complex than the original MAP decoding algorithm. The MAP

32
decoding algorithm was first introduced (in 1974) by Bahl, Cocke, Jelinek and Raviv and

is also known as the BCJR algorithm. However, due to the push for strikingly low bit

error rates, the MAP or the Log-MAP has been most commonly used in irregular turbo

codes since they are based on the optimal decoding rule [1]. In contrast, the SOVA [3] is

an approximation to the MAP sequence decoder and will have a slightly worse bit error

performance. Though SOVA suffers from performance degradation as opposed to the

Log-MAP decoding rules, it has much reduced complexity.

MAP Algorithm

To first understand the decoding of turbo codes, a preliminary understanding of

the MAP (BCJR) algorithm is necessary. In the original paper, the idea was set out to

estimate the a posteriori probabilities of the states and transitions of a Markov sequence

transmitted through a discrete memoryless channel. This work resulted in an algorithm

that minimizes symbol error rates while trying to decode block and trellis codes.

The aim of the MAP algorithm is to minimize the symbol error rate for the

decoding of trellis and block codes. Therefore, after receiving the information through the

channel, the job of the decoder is to determine the most likely input bits

(original/uncoded information sequence), based on the received symbols. Since the input

is over the binary alphabet, it is conventional to form a log-likelihood ratio (LLR) and

base the bit estimates on comparisons based on magnitude of the likelihood ratio to a

threshold. The log-likelihood ratio for the input symbol indexed at time t is defined as

33
 The decoder produces estimates of the information bits based on the values of the log-

likelihood ratio. The magnitude of the log-likelihood ratio is defined as the soft output or

soft value which can be passed after processing to the other decoder as an apriori

information. Also, the sign of the LLR determines the hard estimate of the original

information sequence. The estimator obeys the following rule:

In order to perform the decoding when the information is received through the

channel, the log-likelihood ratio must be computed.

Puncturing

Puncturing is used to delete one or more coded parity bits from a codeword in a

code. This well known technique helps to obtain high-rate codes without modifying the

structure of the encoder and the decoder circuit for an existing encoder/decoder circuit.

Although punctured codes makes a system flexible by allowing the change in the code

rate they usually suffer from performance degradation as compared to the unpunctured

codes.

3.3 Interleaving

The decoding algorithm performs poorly when it is presented with bit errors that

are all bunched together in the stream. Because the sub-carriers are subject to flat fading

bit errors usually do occur in groups when a sub-carrier is in a deep fade. To protect

against this, COFDM uses time interleaving and frequency interleaving.

34
Interleaver Design

An interleaver is a device that mixes up the symbols from several codewords so

that the symbols from any given codeword give a random effect. When the deinterleaver

reconstructs the codeword by arranging the received sequence to its original order, error

bursts introduced by the channel are broken up and spread across several code words.

Interleavers were used practically in both serially concatenated codes and in multipath

fading channels to enhance the overall error correcting capability of the coding scheme.

They were used differently for the first time in a parallel concatenated code and proved to

reduce the number of codewords with small distances in the code distance spectrum (i.e.

generates less code words with minimum hamming distance).

(a) Golden Relative Prime Interleaver

The golden relative prime interleaver computes the scrambler indexes using the

following relation

………….(28)

where, s is an integer starting index, p is an integer index increment, and N is the

interleaver length. The values of N and p must be relative primes in order to ensure the

uniqueness of each index element. The starting index s is usually set to 0, but other

integer values can also be selected. The relative prime increment p is chosen close to one

of the non-integer values of c, which is computed as shown below

…………………..(29)

35
In the expression of c, g is the golden section value, m is any integer greater than zero, r

is the index spacing, and j is any integer modulo r. The preferred values of j, m and r are

0, 1 and 1 respectively.

(b) Golden Interleaver

The Golden interleaver does not depend on the usage of relative primes and

integer modulo arithmetic but is rather based on the principle of sorting real-valued

numbers derived from the golden section value. After computing the value of c as given

by equation (29) the real-valued golden vector e is calculated as follows:

…………..(30)

where, s is any real starting value. After finding the vector e it is possible to find the

index vector z. If the sorted version of the vector e is denoted by a, then the index vector

z and the sorted vector a will be relates as shown below:

……………………….(31)

finally, the golden interleaver indexes are then computed by the following expression:

………………...(32)

Golden interleaver has been used here.

36
3.4 Serial to Parallel Conversion

The input serial data stream is formatted into the word size required for

transmission and shifted into a parallel format. The data is then transmitted in parallel by

assigning each data word to one carrier in the transmission.

3.5 Inverse Fourier Transform

After the required spectrum is worked out, an inverse Fourier transform is used to

find the corresponding time waveform. The guard period is then added to the start of each

symbol. 

3.6 Guard Period

The guard period used was made up of two sections. Half of the guard period time

is a zero amplitude transmission. The other half of the guard period is a cyclic extension

of the symbol to be transmitted. This was to allow for symbol timing to be easily

recovered by envelope detection. However it was found that it was not required in any of

the simulations as the timing could be accurately determined position of the samples. 

After the guard has been added, the symbols are then converted back to a serial time

waveform. This is then the base band signal for the OFDM transmission.

3.7 Modulation

Binary data from a memory device or from a digital processing stream is used as

the modulating (baseband) signal.

37
 The following steps may be carried out in order to apply modulation to the carriers

in OFDM:

 combine the binary data into symbols according to the number of bits/symbol

selected

 convert the serial symbol stream into parallel segments according to the number

of carriers, and form carrier symbol sequences

 apply differential coding to each carrier symbol sequence

 convert each symbol into a complex phase representation

 assign each carrier sequence to the appropriate IFFT bin, including the complex

conjugates.

 take the IFFT of the result

3.8 Channel model

A channel model is then applied to the transmitted signal. The model allows for the

signal to noise ratio, multipath, and peak power clipping to be controlled. The signal to

noise ratio is set by adding a known amount of white noise to the transmitted signal.

Multipath delay spread then added by simulating the delay spread using an FIR filter. The

length of the FIR filter represents the maximum delay spread, while the coefficient

amplitude represents the reflected signal magnitude. 

Generally, for purpose of simulation, two approaches are used to model a mobile

radio channel. The first one deals with the multipath phenomenon and the second one

deals, in addition to multipath, with frequency selectivity [6]. A Rayleigh flat fading

38
channel can be used to simulate the multipath phenomenon. In this case, the received

symbol, (xk, yk), is given by

………….(33)

where (xk, yk) are two independent Rayleigh distributed random variables and (ik, qk) are

the additive white Gaussian noise components. The frequency selectivity can be

introduced by means of intersymbol interference (ISI). In an air-to-ground

communication, the channel is characterized by a direct line of sight (LOS) and multipath

component with a relative delay [4]. This delayed path creates some ISI on the received

signal

……………………….(34)

with n representing the relative delay of the diffuse component. This model fits the

Rician fading channel.

For applications in microwave LOS radio channels, a similar model has been considered,

the Rummler channel [5] which is a two path model

……………….(35)

where a is the overall attenuation, and the term within parenthesis represents the

interference between two rays having a relative delay and producing a minimum

amplitude at the notch frequencyf0. b is the relative amplitude of the multipath ray. With

39
high speed transmission, the mobile radio channel can be considered as quasi-stationary;

this means that for a short period of time, the channel’s physical parameters do not

change. The channel impulse response corresponds to a sum of echoes with random

arrival times and random amplitudes with a given power profile. Over the period of

stationarity, these parameters can be considered fixed. In such a situation, the channel can

be considered as a fixed multipath fading one. With this hypothesis, we can use

Rummler’s arguments to show that a two ray channel model fits quite well to the physical

situation .The outputs of the channel are coherently demodulated and the soft inputs of

the decoder are then (Xk,Yk).

3.9 Receiver

The receiver basically does the reverse operation to the transmitter. The guard period

is removed. The FFT of each symbol is then taken to find the original transmitted

spectrum. The phase angle of each transmission carrier is then evaluated and converted

back to the data word by demodulating the received phase. The data words are then

combined back to the same word size as the original data.

40
 CHAPTER 4

SIMULATION RESULTS AND DISCUSSION

This work is the first step of a more important project. Given the fact that

interleaving is already exploited by turbo codes, no attention is paid to time or frequency

interleavers as normally used in COFDM schemes. Furthermore, no equalization or any

channel corrections are performed. The simulations have been carried out using

MATLAB. The BER was computed after 3 iterations of the decoder. The number of

simulations is defined to insure that the precision of the BER estimation is less than 1%

with a confidence interval less than 5%. The system parameters are chosen for a

particular real situation where some transmission speed and bandwidth must be respected.

Results have been obtained for both Additive White Gaussian Noise (AWGN) and

Rayleigh fading channel.

Comparison is made between turbo code and 5 %, 10%, and 15% irregular turbo

code and uncoded system. Plots clearly show that irregularity (5 %, 10%, and 15%) gives

improvements of approximately 0.8 dB at BER of 10 -4 in AWGN channel. Irregularity

also gives better performance in Rayleigh channel. Irregularity achieves gain of

approximately 0.8 dB at BER of 10-4 in Rayleigh channel (fig 4.4). There is very slight

improvement in performance from 5 % irregularity to 10 % irregularity and from 10 %

irregularity to 15 % irregularity. It can be seen that the BER has been considerably

reduced by performing forward error correction coding using irregular turbo codes. It can

also be seen that the intersymbol interference due to multipath has been reduced.

41
Figure 4.1: BER performance of 16-QAM OFDM without FEC in AWGN channel

Figure 4.2: BER performance of 16-QAM COFDM in AWGN channel

42
Figure 4.3: BER performance of 16-QAM OFDM without FEC in RAYLEIGH channel

Figure 4.4: BER performance of 16-QAM COFDM in RAYLEIGH channel

43
 CONCLUSIONS
This project presented some new results on the behavior of COFDM using

irregular turbo codes. It was shown that with a simple architecture the system offers quite

acceptable performances. A problem, although difficult, but very interesting to consider,

is the design of good irregular turbo codes for mobile channels. Finally, more

sophisticated mobile radio channels should be considered.

BIBLIOGRAPHY

44
1. S. Goff, A. Glavieux, and C. Berrou, “Turbo-codes and high spectral efficiency

modulation,”Proc, IEEE Int. Conf. on Commun., pp. 645-9, May 1994.

2. P. Robertson and T. Worz, “Coded modulation scheme employing turbo codes,”

Electronics Letters, vol. 31, pp.1546-7, Aug. 31st, 1995.

3. J. Hagenauer and L. Papke, “Decoding turbo codes with the soft-output Viterbi

algorithm (SOVA),” Proc, IEEE Int. Symp. on Inform. Theory, p. 164, June 1994.

4. J. Proakis, “Digital Communications”, third edition. New York: McGraw-Hill, Inc.,

1995.

5. W. D. Rummler et al., “Multipath fading channels models for microwave digital

radio”, IEEE Communication Magazine, vol. 24, Nov. 1996.

6. W. C. Lee, “Mobile Communication Engineering”, McGraw-Hill, 1982.

7. Simon Haykin “Communication Systems”, John Willey & sons, 4th edition

8. S. B. Weinstein, “Data Transmission by Frequency-Division Multiplexing Using the

Discrete Fourier Transform”, IEEE Trans. Commun. vol. COM-19, no. 5, pp. 628-3,

1971.

9. The how and why of COFDM, J. H. Stott,

1998,www.bbc.co.uk/rd/pubs/papers/pdffiles/ptrev_278-stott.pdf

10. Orthogonal Frequency Division Multiplexing (OFDM) Explained, Magis Networks,

Inc., February 8, 2001, www.magisnetworks.com

45