INTRODUCTION

The fundamental problem of communication is that of reproducing at one point either

exactly or approximately a message selected at another point. Frequently the messages have
meaning that is they refer to or are correlated according to some system with certain
physical or conceptual entities. These semantic aspects of communication are irrelevant to
the engineering problem. The significant aspect is that the actual message is one selected
from a set of possible messages. The system must be designed to operate for each possible
selection, not just the one which will actually be chosen since this is the information
conveyed by each possible message, which can be large, provided it belongs to a finite set
of messages. This would mean that the number of messages generated by the information
source should be finite, leading to the development of what is known as a digital
communication system.

There are various performance characteristics to study the performance of various digital
systems on a comparative basis, some of which are error performance (Eb/N0 vs. SNR) and
transmission rate. Usually, it is not possible to optimize these parameters without degrading
the remainder of the performance specifications. Hence, usually a tradeoff is achieved
between these performance specifications.

For about five decades coding theorists have been looking in vain for codes capable of
approaching the Shannon limit. From this unsuccessful search arises the theorem : All
codes are good, except those that we know of. The study is more complicated though, since
finding good codes is a simple task. Indeed, randomly generated codes with large block
sizes will be very good with high probability, the problem lies in the fact that, while
encoding is always a rather simple task, the decoding complexity for a randomly generated
code increases exponentially with the block size, and thus quickly becomes unmanageable.
Thus, the previously mentioned theorem should be rephrased as: all codes are good, except
those that we know how to decode. The history of the past fifty years of coding theory is
about the struggle of conjugating a great structure in the code architecture to facilitate
decoding with a random look in the code words distribution to enhance the code strength
according to the message implicit in the proof of Shannon coding theorem.

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In the last decade, coding emphasis has been in the area of Convolutional codes since in
almost every application, Convolutional codes outperform block codes for the same
implementation complexity of the encoder-decoder. The notion of state and the existence of
efficient sequential decoding algorithms make convolutional codes easily adaptable for use
with decoders that exploit channel memory.

Reliable transfer of information by communication systems would require robustness

towards noise effects introduced by non-ideal properties of the communication channel.
The error performance (Eb/N0) of the communication system, therefore, becomes a major
parameter of interest and is recorded and compared for different configurations of the
communication system. By configuration, here, is meant the decoding strategy that is in
use. High fidelity values for this parameter (i.e., low BER) can be achieved through use of
optimum decoding algorithms for convolutional codes (e.g. Viterbi algorithm). However,
this remains true only for the case where the channel is affected by statistically independent
noise. For such channels, Viterbi algorithm is usually used to implement the maximum-
likelihood sequence detection (MLSD) approach that represents the optimum detection
strategy for data signals.

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CHAPTER 1

OFDM

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 OFDM

1.1 SINGLE CARRIER APPROACH OF TRANSMISSION:

In a conventional serial data system data bits are transmitted serially. Therefore high data
bit rate is required. A single carrier system modulates information onto one carrier using
frequency, phase, or amplitude adjustment of the carrier. However each carrier acquires
total available spectrum. As higher bandwidths (data rates) are used, the duration of one bit
or symbol of information becomes smaller. The system then becomes more susceptible to
loss of information due to fading, delay spread and Doppler shift.

1.2 PARALLEL AND MULTICARRIER APPROACH:

A parallel data system eliminates many problems encountered with serial systems. A
parallel system is one in which several sequential streams of data are transmitted parallely
so that at any instant many data elements are being transmitted. Over different sub-carriers.
Each sub-carrier contains a lower bit rate, which makes the system less sensitive to physical
channel’s delay spread, while maintaining the original transmission rate. In such a system
the spectrum of an individual data element occupies only a small part of the available
bandwidth .therefore spectral efficiency is increases.
OFDM is a example of parallel data system.

1.3 OVERVIEW OF OFDM:

In the past, amplitude modulation (AM) and frequency modulation (FM) were the most
common ways of transmitting information by radio. This was because the circuits to
transmit and receive these signals were relatively simple. Now with the availability of very
cheap and powerful digital signal processors (DSP’s) much more mathematically
sophisticated systems are possible. One of them is orthogonal frequency division
multiplexing (OFDM).

OFDM represents a different system-design approach. It can be thought of as a

combination of modulation and multiple-access schemes that segments a communications
channel in such a way that many users can share it. Whereas TDMA segments are
according to time and CDMA segments are according to spreading codes, OFDM segments
are according to frequency. It is a technique that divides the spectrum into a number of
equally spaced tones and carries a portion of a user's information on each tone. A tone can

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be thought of as a frequency, much in the same way that each key on a piano represents a
unique frequency. OFDM can be viewed as a form of frequency division multiplexing
(FDM), however, OFDM has an important special property that each tone is orthogonal
with every other tone. FDM typically requires there to be frequency guard bands between
the frequencies so that they do not interfere with each other. OFDM allows the spectrum of
each tone to overlap, and because they are orthogonal, they do not interfere with each other.
By allowing the tones to overlap, the overall amount of spectrum required is reduced.

Fig.1.1 OFDM Tones

OFDM is a modulation technique in that it enables user data to be modulated onto the
tones. The information is modulated onto a tone by adjusting the tone's phase, amplitude, or
both. In the most basic form, a tone may be present or disabled to indicate a one or zero bit
of information; however, either phase shift keying (PSK) or quadrature amplitude
modulation (QAM) is typically employed. An OFDM system takes a data stream and splits
it into N parallel data streams, each at a rate 1/N of the original rate. Each stream is then
mapped to a tone at a unique frequency and combined together using the inverse Fast
Fourier transform (IFFT) to yield the time-domain waveform to be transmitted.

For example, if a 100-tone system were used, a single data stream with a rate of 1 megabit
per second (Mbps) would be converted into 100 streams of 10 kilobits per second (kbps).
By creating slower parallel data streams, the bandwidth of the modulation symbol is
effectively decreased by a factor of 100, or, equivalently, the duration of the modulation
symbol is increased by a factor of 100. Proper selection of system parameters, such as the

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number of tones and tone spacing, can greatly reduce, or even eliminate, ISI, because
typical multipath delay spread represents a much smaller proportion of the lengthened
symbol time. Viewed another way, the coherence bandwidth of the channel can be much
smaller, because the symbol bandwidth has been reduced. The need for complex multi-tap
time-domain equalizers can be eliminated as a result.

1.4 ORTHOGONALITY:

The word orthogonal indicates that there is a precise mathematical relationship between the
frequencies of the carriers in the system. In a normal frequency-division multiplex system,
many carriers are spaced apart in such a way that the signals can be received using
conventional filters and demodulators. In such receivers, guard bands are introduced
between the different carriers and in the frequency domain, which results in a lowering of
spectrum efficiency.

It is possible, however, to arrange the carriers in an OFDM signal so that the sidebands of
the individual carriers overlap and the signals are still received without adjacent carrier
interference. To do this, the carriers must be mathematically orthogonal.

The receiver acts as a bank of demodulators, translating each carrier down to DC, with the
resulting signal integrated over a symbol period to recover the raw data. If the other carriers
all beat down the frequencies that, in the time domain, have a whole number of cycles in
the symbol period T, then the integration process results in zero contribution from all these
other carriers. Thus, the carriers are linearly independent (i.e., orthogonal) if the carrier
spacing is a multiple of 1/T.

Fig.1.2 Spectra of (a) an OFDM subchannel and (b) and OFDM signal.

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Figure (a) shows the spectrum of the individual data of the subchannel. The OFDM signal,
multiplexed in the individual spectra with a OFDM for frequency spacing be equal to the
transmission speed of each subcarrier, is shown in Figure (b). Above figure shows that at
the center frequency of each subcarrier, there is no crosstalk from other channels.
Therefore, if we use DFT at the receiver and calculate correlation values with the center of
frequency of each subcarrier, we recover the transmitted data with no crosstalk. In addition,
using the DFT-based multicarrier technique, frequency-division multiplex is achieved not
by bandpass filtering but by baseband processing.

1.4.1 Condition For Orthogonality:

Function Xn(t) & Xm(t) are said to be orthogonal with respect to each other over the
interval a<t<b if they satisfy the following condition,

Int(a,b) Xn(t).X*m(t) dt = 0 where n is not = m

=k where n = m

The orthogonality of the carriers means that each carrier has an integer number of cycles
over a symbol period. The result of orthogonality is no interference between the carriers,
allowing them to be spaced as close as theoretically possible.

Frequency spacing = 1 / symbol period

The carriers have common, precisely-chosen frequency spacing. This is the inverse of the
duration, called the active symbol period, over which the receiver will examine the signal,
performing the equivalent of an 'integrate-and-dump' demodulation. This choice of carrier
spacing ensures orthogonality (the 'O' of OFDM) of the carriers -- the demodulator for one
carrier does not 'see' the modulation of the others, so there is no crosstalk between carriers,
even though there is no explicit filtering and their spectra overlap.

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1.5 OFDM BLOCK DIAGRAM:

Fig.1.3 OFDM block diagram

OFDM TRASNMITTER

• Random data generator:

Random data generator generates data to be transmitted .If this data is not in electrical form
then 1st it is converted into electrical signal. This signal is known as base band signal.

• Serial to Parallel Conversion:

The input serial data stream is formatted into the word size required for transmission, e.g. 2
bits/word for QPSK, and shifted into a parallel format. The data is then transmitted in
parallel by assigning each data word to one carrier in the transmission.

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 • Modulation of Data:
The data to be transmitted on each carrier is then differential encoded with previous
symbols, then mapped into a Phase Shift Keying (PSK) format. Since differential encoding
requires an initial phase reference an extra symbol is added at the start for this purpose. The
data on each symbol is then mapped to a phase angle based on the modulation method. For
example, for QPSK the phase angles used are 0, 90, 180, and 270 degrees. The use of phase
shift keying produces a constant amplitude signal and was chosen for its simplicity and to
reduce problems with amplitude fluctuations due to fading.

• 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.
The idea behind the analog implementation of OFDM can be extended to the digital
domain by using the discrete Fourier Transform (DFT) and its counterpart, the inverse
discrete Fourier Transform (IDFT). These mathematical operations are widely used for
transforming data between the time-domain and frequency-domain. These transforms are
interesting from the OFDM perspective because they can be viewed as mapping data onto
orthogonal subcarriers. For example, the IDFT is used to take in frequency-domain data
and convert it to time-domain data. In order to perform that operation, the IDFT correlates
the frequency-domain input data with its orthogonal basis functions, which are sinusoids at
certain frequencies. This correlation is equivalent to mapping the input data onto the
sinusoidal basis functions.

In practice, OFDM systems are implemented using a combination of fast Fourier Transform
(FFT) and inverse fast Fourier Transform (IFFT) blocks that are mathematically equivalent
versions of the DFT and IDFT, respectively, but more efficient to implement. An OFDM
system treats the source symbols (e.g., the QPSK or QAM symbols that would be present in
a single carrier system) at the transmitter as though they are in the frequency-domain.
These symbols are used as the inputs to an IFFT block that brings the signal into the time-
domain. The IFFT takes in N symbols at a time where N is the number of sub carriers in the
system. Each of these N input symbols has a symbol period of T seconds. Recall that the
basis functions for an IFFT are N orthogonal sinusoids. These sinusoids each have a
different frequency and the lowest frequency is DC. Each input symbol acts like a complex

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weight for the corresponding sinusoidal basis function. Since the input symbols are
complex, the value of the symbol determines both the amplitude and phase of the sinusoid
for that subcarrier. The IFFT output is the summation of all N sinusoids. Thus, the IFFT
block provides a simple way to modulate data onto N orthogonal subcarrier. The block of N
output samples from the IFFT make up a single OFDM symbol. The length of the OFDM
symbol is NT, where T is the IFFT input symbol period mentioned above.

After some additional processing, the time-domain signal that results from the IFFT is
transmitted across the channel. At the receiver, an FFT block is used to process the received
signal and bring it into the frequency-domain. Ideally, the FFT output will be the original
symbols that were sent to the IFFT at the transmitter. When plotted in the complex plane,
the FFT output samples will form a constellation, such as 16-QAM. However, there is no
notion of a constellation for the time-domain signal. When plotted on the complex plane,
the time-domain signal forms a scatter plot with no regular shape. Thus, any receiver
processing that uses the concept of a constellation (such as symbol slicing) must occur in
the frequency-domain.

• 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.

• Cyclic prefix:
A major problem in most wireless systems is the presence of a multipath channel. In a
multipath environment, the transmitted signal reflects off of several objects. As a result,
multiple delayed versions of the transmitted signal arrive at the receiver. The multiple
versions of the signal cause the received signal to be distorted. Many wired systems also
have a similar problem where reflections occur due to impedance mismatches in the
transmission line.

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A multipath channel will cause two problems for an OFDM system. The first problem is
intersymbol interference. This problem occurs when the received OFDM symbol is
distorted by the previously transmitted OFDM symbol. The effect is similar to the
intersymbol interference that occurs in a single-carrier system. However, in such systems,
the interference is typically due to several other symbols instead of just the previous
symbol; the symbol period in single carrier systems is typically much shorter than the time
span of the channel, whereas the typical OFDM symbol period is much longer than the time
span of the channel. The second problem is unique to multicarrier systems and is called
Intrasymbol Interference. It is the result of interference amongst a given OFDM symbol's
own subcarrier.

The orthogonality of subchannels in OFDM can be maintained and individual subchannels

can be completely separated at the receiver when there are no ISI and ICI introduced by
transmission channel distortion. In practice, this condition cannot be obtained. Since the
spectra of an OFDM signal is not strictly band limited (sinc(f)function), as multipath causes
each subchannels to spread energy into the adjacent channels and consequently cause ISI.
A simple solution is to increase symbol duration or the number of carriers so that distortion
becomes insignificant. However, this method may be difficult to implement in terms of
carrier stability, Doppler shift, FFT size and latency.

In OFDM, cyclic preffix is used to avoid interference. The cyclic prefix is actually a copy
of the last portion of the data symbol appended to the front of the symbol during the guard
interval, as shown in Figure. Multipath causes tones and delayed replicas of tones to arrive
at the receiver with some delay spread. This leads to misalignment between sinusoids,
which need to be aligned as in Figure to be orthogonal. The cyclic prefix allows the tones
to be realigned at the receiver, thus regaining orthogonality.

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 Fig.1.4 Cyclic Prefix

The total symbol duration is,

Ttotal= Tg + T
Where, Tg : Guard interval.
T : Useful symbol duration,

When the Tg is longer than channel impulse response or multipath delay ISI can be
eliminated. (Tg/T ) is application dependent. Since the insertion of guard interval will
reduce data throughput, usually Tg < (T/4).

At the receiver, certain position within the cyclic prefix is chosen as the sampling starting
point, which satisfies the criteria,

Where τ max is the worst-case multi-path spread. As illustrated in the following figure,
once the above condition is satisfied, there is no ISI since the previous symbol will only
have effect over samples within [0, τ max ] .

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It is also clear from the figure that sampling period starting from Tx will encompass the
contribution from all the multi-path components so that all the samples experience the same
channel and there is no ICI

Fig.1.5 Condition of Cyclic Prefix

 OFDM CHANNEL:
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.

 OFDM 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.

SUMMARY:

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In this chapter we had discussed why we go for OFDM system. We had an overview about
orthogonality concept. Finally the process of OFDM was represented through a block
diagram.

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 CHAPTER 2

CHANNEL CODING

CHANNEL CODING

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INTRODUCTION:
Channel Coding refers to the class of signal transformations designed to improve
communications performance by enabling the transmitted signals to better withstand the
effects of various channel impairments, such as noise, interference and fading. These
signal-processing techniques can be thought of as vehicles for accomplishing desirable
system trade-offs (e.g. Error-performance versus Bandwidth, Power versus Bandwidth).
The use of large-scale integrated circuits (LSI) and high-speed digital signal processing
(DSP) techniques have made it possible to provide as much as 10 dB performance
improvement through these methods, at much less cost than through the use of most other
methods such as higher power transmitters or larger antennas.

Since in every application, convolutional codes outperform block codes for the same
implementation complexity of the encoder-decoder. There are two more significant
advantages associated with convolutional codes, which make them easily adaptable for use
with decoders that exploit the channel memory without increasing the concomitant
complexity.

2.1 STRUCTURE OF A CONVOLUTIONAL CODE AND ITS CODE

PARAMETERS:
A convolutional code is described by three integers, n, k and K, where the ratio k/n has the
same code rate significance (information per coded bit) that it has for block codes;
however, n does not define a block or codeword length as it does for block codes. The
integer K is a parameter known as constraint length; it represents the number of k-tuple
stages in the encoding shift register. An important characteristic of convolutional codes,
different from block codes, is that an encoder has memory – the n-tuple emitted by the
convolutional encoding procedure is not only a function of an input k-tuple, but is also a
function of the previous K-1 input k tuples. In practice, n and k are small integers and K is
varied to control the capability and complexity of the code.

2.2 CONNECTION REPRESENTATION OF A CONVOLUTION CODE:

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The convolutional encoder used in the project is as shown in the figure below. The figure
illustrates a ( 1, 2) convolutional encoder with a constraint length K = 3. There are n = 2
modulo-2 adders; thus the code rate k/n is ½.

Fig.2.1 Convolution Encoder

At each input bit time, a bit is shifted into the leftmost stage and the bits in the register are
shifted one position to the right. Next, the sample switch samples the output of each
modulo-2 adder (i.e., first the upper adder, then the lower adder), thus forming the code
symbol pair making up the branch word associated with the bit just inputted. The sampling
is repeated for each inputted bit. The choice of connections between the adders and the
stages of the registers gives rise to the characteristics of the code. Any change in the choice
of connections results in a different code. The connections are not chosen or arranged
arbitrarily.

Unlike a block code that has a fixed word length n, a convolutional code has no particular
block size. However, convolutional codes are often forced into a block structure by periodic
truncation. This requires a number of zero bits to be appended to the end of the input data
sequence, for the purpose of clearing or flushing the encoding shift register of the data bits.
Since the added zeros carry no information, the effective code rate falls below k/n. To keep
the code rate close to k/n, the truncation period is generally made as long as practical.

2.3 IMPULSE RESPONSE OF A CONVOLUTIONAL ENCODER:

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One can approach the encoder in terms of its impulse response – that is, the response of the
encoder to a single “one” bit that moves through register in the above it. Consider the
contents of the figure as a one moves through it:

Input sequence: 1 0 0
Output sequence: 1 1 1 0 1 1

The output sequence for the input “one” is called the impulse response of the encoder.
Then, for the input sequence m = 1 0 1, the output may be found by the superposition or
linear addition of the time-shifted input “impulses” as follows:

Observe that this is the same output as that obtained in figure 2.1, demonstrating that
convolutional codes are linear. It is from this property of generating the output by linear
addition of time-shifted impulses or the convolution of the input sequence with the impulse
response of the encoder, that it derives the name Convolutional encoder.

2.4 STATE REPRESENTATION:

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A convolutional encoder belongs to the class of devices known as finite state machine,
which is the general name, given to machines that have a memory of past signals. The
adjective finite refers to the fact that there are only a finite number of unique states that the
machine can encounter. In the most general sense, the state consists of the smallest amount
of information that, together with a current input to the machine, can predict the output of
the machine. The state provides some knowledge of the past signaling events and the
restricted set of possible outputs in the future. For a rate 1/n convolutional encoder, the
state is represented by the contents of (K-1) stages. Knowledge of the state together with
knowledge of next input is necessary and sufficient to determine the next output.

2.4.1 STATE DIAGRAM:

One way to represent simple encoders is with a state diagram; such a representation for the
encoder is shown in the figure below. The states, shown in the boxes of the diagram
represent the possible contents of the right most K-1 stages of the register, and the paths
between the states represent the output branch words resulting from such state transitions.
The states of the register are designated a = 00, b = 10, c = 01 and d = 11. The diagram
shown in the figure below illustrates all the state transitions that are possible for the
encoder in figure. There are only two transitions emanating from each state, corresponding
to the possible input bits. Next to each path between states is written the output branch
word associated with the state transition. Notice that it is not possible in a single transition
to move from a given state to any arbitrary state. For example, if the present encoder state
is 00, the only possibilities for the state at the next shift are 00 or 10.

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 Fig.2.2 Encoder State Diagram

.
2.4.2 TREE DIAGRAM:
Although the state diagram completely characterizes the encoder, one cannot easily use it
for tracking the encoder transitions as a function of time since the diagram cannot represent
time history. The tree diagram adds the dimension of time to the state diagram. The tree
diagram for the convolutional encoder shown in figure 2.1is illustrated in figure. At each
successive input bit time, the encoding procedure can be described by traversing the
diagram from left to right, each tree branch describing an output branch word. The
branching rule for finding a code word sequence is as follows: If the input bit is a 0, its
associated branch word is found by moving to the next right most branch in the upward
direction. If the input bit is a 1, its branch word is found by moving to the next right most
branch in the downward direction. Assuming that the initial contents of the encoder is all
zeros, the diagram shows that if the first input bit is a zero, the output branch word is 00
and , if the first input bit is a one, the output branch word is 11. Following this procedure,
we see that the input sequence 1 1 0 0 1 1 traces the heavy line drawn on the tree diagram.
This path corresponds to the output code word sequence 1 1 0 1 0 1 0 0 0 1. The number of
branches increases as a function of 2L, where L is the number of branch words in the
sequence.

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21
2.4.3 TRELLIS DIAGRAM:
After third branching, the upper and lower halves of the tree diagram shown above are
identical. This means that any two nodes having the same state label at the same time ti can
be merged, since all succeeding paths will be indistinguishable. If we do this to the tree
structure shown above, we obtain another diagram, called the trellis diagram. The Trellis
diagram, by exploiting the repetitive structure, provides a more manageable encoder
description than does the tree diagram. The trellis diagram for the convolutional encoder of
figure 2.1, is shown in figure 2.4.

SUMMARY:

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For forward error detection and correction in OFDM, we make use of convolutional
encoders with suitable code rates, The process of encoding involves the use of the impulse
response method, state diagram, tree diagram and the trellis diagram.

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 CHAPTER 3

INTERLEAVING
&
EQUALIZING

IN DATA TRANSMISSION

3.1 INTERLEAVING IN DATA TRANSMISSION:

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Today, interleaving is mainly used in digital data transmission technology, to protect the
transmission against burst errors.These errors overwrite a lot of bits in a row, but seldom
occur. Interleaving is used to solve this problem. All data is transmitted with some control
bits (independently from the interleaving), such as error correction bits, that enable the
channel decoder to correct a certain number of altered bits. If a burst error occurs, and more
than this number of bits is altered, the codeword cannot be correctly decoded. So the bits of
a number of codewords are interleaved and then transmitted. That way, a burst error affects
only a correctable number of bits in each codeword, so the decoder can decode the
codewords correctly.
Also, the complexity, required cost and effort for error correction mechanism of the
communication can be reduced by employing interleaving method since that the data errors
can be controlled to reasonable range.
Let's take a look at an example. We apply an error correcting code so that the channel
codeword has four bits and one-bit errors can be corrected. The channel codewords are put
into a block like this: aaaabbbbccccddddeeeeffffgggg.

e.g: Consider transmission without interleaving:

Error-free message: aaaabbbbccccddddeeeeffffgggg
Transmission with a burst error: aaaabbbbccc____deeeeffffgggg

The codeword dddd is altered in three bits, so it cannot be decoded (decoding failure) or
might be decoded into a wrong codeword. Which of the two happens depends on the error
correcting code applied.

Now, let's do the same with interleaving:

Error-free transmission: aaaabbbbccccddddeeeeffffgggg
Interleaved: abcdefgabcdefgabcdefgabcdefg
Transmission with a burst error: abcdefgabcd____bcdefgabcdefg
Deinterleaved with a burst error: aa_abbbbccccdddde_eef_ffg_gg

In each of the codewords aaaa, eeee, ffff, gggg, only one bit is altered, so our one-bit-error-
correcting-code can decode everything correctly.
For one more example, if we got a meaningful sentence like:
ThisIsAExampleForInterleaving.

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And, if the correction mechanism of the communication can correct maximum 7-bit burst
errors. Let us see the sentence without interleaving.

Consider transmission without interleaving:

Error-free transmission: ThisIsAExampleForInterleaving

transmission with a burst error: ThisIsA_______ForInterleaving

We can find that the word "Example" is lost and can not be recovered.
Same as above, we do it again with interleaving.

Consider transmission with interleaving:

Error-free transmission: TixlreaghsaeIrviAmFnlisEpoten
transmission with a burst error: TIxlrea_______viAmFnlisEpoten
De-interleaving: T_isI_AEx_mpl_For_nte_leavin_

No single word is completely lost and it is easy to recover them.

Of course, latency is added by this, because we cannot send the second bit of codeword a
before awaiting the first bit of codeword gggg.

Thus,interleaving improves the performance of digital radio systems at the cost of

increasing memory space requirements, system complexity and time-delay.

The mwmory space increase is normally tolerable. The complexity issue is relative and it is
always decreasing due to advances in technology. However, time-delay increase may
render interleaving impractical in certain applications(e.g. Voice communications).

Disadvantages of interleaving:
Latancy is increased twofold when interleaving is used to improve error performance. This
is because the entire interleaved block must be received before the critical data can be
returned.

EQUALIZATION:

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In the transmission of digital information at high data rates over telephone channels. It is
well known that the channel distorts the signal and causes inter symbol interference (ISI)
among the data symbols. The ISI may cause errors when we attempt to recover the data. If
the data are transmitted over the direct dial network, the channel characteristics that cause
the distortion vary considerably form one telephone call to another. Therefore, it is
appropriate to assume that the channel, which is well modeled as a linear system, is known
to the receiver that much recover the digital information. In such a case the problem is to
design a corrective system which, when cascaded with the original system, produces on
output that corrects for the distortion caused by the channel and thus yields a replica of the
desired transmitted signal. In digital communications such a corrective system is called an
equalizer (Equalizer is called an inverse system in linear system theory)
The Process of correcting the channel-induced distortion is called equalization. The filter
used to perform such a process is called an equalizer.

Tapped – delay line equalizer:

A device well suited for the design of a linear equalizer is the tapped – delay line equalizer
(transversal filter). For symmetry. The total number of taps Is chosen to be (2N +1), with
the weights denoted by Wo, W1,…Wn. The impulse response of the tapped – delay – line
equalizer is therefore.

N
H(t)= ∑ Wk δ ( t-KT)
K= -N

Where δ (t) is the delta function and the delay T is chosen equal to the symbol duration.
Suppose that the tapped – delay – line equalizer is connected in cascaded with a linear
system whose impulse response is c (t). Let p (t) denote the impulse response of the
equalized system.

Impulse response
p (t)

Linear System Tapped – delay line

c (t) Equalizer h (t)

Cascade connection of linear system and tapped – delay line equalizer

27
 Tapped – delay – line filter

Adaptive Equalization:
In a telecommunication environment the channel is usually time varying. So the use of
fixed equalizer designed on the basis of average channel characteristics may not adequately
reduce ISI. To realize the full transmission capability of a telephone channel there is need
for adaptive equalization. The process of equalization is said to be adaptive when the
equalizer adjust itself continuously and automatically by operating on the input signal.

Two methods
1. Pre channel equalization – feedback path is required (equalization takes place at the
transmitting side)
2. Post channel equalizer
- At the receiving side
- Prior to data transmission, a training sequence is transmitted through the channel so as
to adjust the filter parameters to optimum values. Equalizer is positioned after the
receiver filter at the receiving side.

28
Elements of an adaptive filter:
1. Prior to data transmission a known test sequence called training sequence is
transmitted on the channel.
2. Resulting response ‘Yn’ is obtained in the receiver by measuring output of the
transversal filter at the sampling instants.
3. An error sequence ‘en’ is obtained by subtracting the received response sequence
from the desired response sequence ‘an’
En = an – yn
4. The Error sequence ‘en’ is used to determine the gain coefficients. An algorithm
(LMS, Least mean square algorithm) is used for optimum setting of the coefficients.
Adaptive equalizers are provided in high-speed modems.

Summary:
Interleaving & equalizing enhances the quality of digital transmission over the radio fading
channel. This enhancement comes at the cost of introducing additional time-delay, memory
space requirement and system complexity.

29
 CHAPTER 4

MODULATION

MODULATION

30
INTRODUCTION:
The digital signal is in the form of pulses. When these pulses are to be transmitted over a
communication medium without distorting its shape, the communication medium must
have a large bandwidth. Thus it is necessary to convert the digital data into some kind of
analog signal so that the bandwidth requirement is reduced. For this conversion we make
use of certain modulation techniques.

4.1 QUADRATURE AMPLITUDE MODULATION:

Quadrature amplitude modulation (QAM) is a modulation scheme in which two sinusoidal
carriers, one exactly 90 degrees out of phase with respect to the other, are used to transmit
data over a given physical channel. Because the orthogonal carriers occupy the same
frequency band and differ by a 90 degree phase shift, each can be modulated
independently, transmitted over the same frequency band, and separated by demodulation
at the receiver. For a given available bandwidth, QAM enables data transmission at twice
the rate of standard pulse amplitude modulation (PAM) without any degradation in the bit
error rate (BER). QAM and its derivatives are used in both mobile radio and satellite
communication systems.

4.2 BLOCK DIAGRAM:

 TRANSMITTER:

The following picture shows the ideal structure of a QAM transmitter, with a carrier
frequency f0 and Ht the frequency response of the transmitter's filter:

fig.4.1 QAM TRANSMITTER

31
First the flow of bits to be transmitted is split into two equal parts: this process generates
two independent signals to be transmitted. They are encoded separately. Then one channel
(the one "in phase") is multiplied by a cosine, while the other channel ("in quadrature") is
multiplied by a sine. This way there is a phase of 90° between them. They are simply added
one to the other and sent through the real channel.

The sent signal can be expressed in the form:

where vc[n] and vs[n] are the voltages applied in response to the nth symbol to the cosine and
sine waves respectively.

 RECEIVER:
The receiver simply performs the inverse process of the transmitter. Its ideal structure is
shown in the picture below with Hr the receive filter's frequency response:

fig.4.2 QAM RECEIVER

Multiplying by a cosine (or a sine) and by a low-pass filter it is possible to extract the
component in phase (or in quadrature). Then the two flows of data are merged back. In any
application, the low-pass filter will be within hr (t).

32
SUMMARY:
We make use of QAM modulation to transmit the data bits more efficiently. At the receiver
side , the performance of the particular system can be evaluated using certain parameters
such as the bit error rate ,keeping SNR constant.

33
 CHAPTER 5

CHANNELS

34
 CHANNELS

INTRODUCTION:
A channel can describe everything from the source to the sink of a radio signal. However,
the aspect of this channel that lies beyond the control of communication system design
actually comes into act in between the transmitter output and receiver input. The manner in
which the channel introduces the error process provides the basis for classification of
channels.

The most general forms of channels fall in one of the two categories:
1. Channels without memory,
2. Channels with finite states or with memory.

Among the two, the errors caused by the memoryless channels are statistically independent.
However the finite state channel introduces errors that are statistically dependent and
sometimes lead to burst errors.

5.1 MEMORYLESS CHANNEL:

The memoryless channel models that could be considered for appraising the performance
of the communication system include: Discrete memoryless channel (DMC), binary
symmetrical channel (BSC) and Gaussian channel.

 GAUSSION CHANNEL:

It is a channel with a discrete input alphabet and a continuous output alphabet over the
range . The channel adds noise to the symbols. Since the noise is a Gaussian
random variable with zero mean and variance 1ð, the resulting probability density function
(pdf) of the received random variable z, conditioned on the symbol uk, can be

35
For all z, where k = 1,2,…, M. When the demodulator output consists of a continuous
alphabet or its quantized approximation (with greater than 2 quantization levels) the
demodulator is said to make soft decisions.

5.2 MEMORY CHANNEL:

A channel that has memory is one that exhibits mutually dependent signal transmission
impairments. A channel that exhibits multipath fading, where signal arrive at the receiver
over two or more paths of different lengths is an example of a channel with memory. The
effect is that the signals can arrive out of phase with each other, and the cumulative
received signal is distorted. Wireless mobile communication channels, as well as
ionospheric and tropospheric propagation channels, suffer from such phenomena. Also,
some channels suffer from switching noise and other burst phenomena. All of these time
correlated impairments result in statistical dependence among successive symbol
transmissions. That is, the disturbances tend to cause errors that occur in bursts, instead of
isolated events.

Under the assumption that the channel has memory, the errors can no longer be
characterized as single randomly distributed bit errors whose occurrence is independent
from bit to bit. Most convolutional codes are designed to combat random independent
errors. The result of a channel having memory on such coded signals is to cause
degradation in error performance. Coding techniques for channels with memory have been
proposed, but the greatest problem with such coding is the difficulty in obtaining accurate
models of the often time-varying statistics of such channels. Hence, the modeling of these
bursty channels needs to be done to aid their analysis, which is available in.

SUMMARY:
We saw that memoryless channels are unaffected by error bursts while memory channels
are affected by burst noise.

36
 CHAPTER 6

CHANNEL DECODING

37
 CHANNEL DECODING

INTRODUCTION:
The function block is to decode the original data sequence from the received information
that is received at the receiver and hence its efficiency is of paramount importance for
improved performance of the digital communication system. Consequently, under the
project, greatest emphasis is being laid on data processing after information has been
corrupted by noise i.e. the stage where channel decoding comes in to act. With respect to
the two different kinds of channels, we consider two decoding strategies: one, that doesn’t
explore it’s channel memory by virtue of the interleaving it uses and other that makes use
of the channel memory with the effect that it is referred to as memory decoding. However,
the algorithms for decoding remain broadly the same for both. The various algorithms for
decoding fall in one of the two categories:
1. Maximum likelihood decoding – Viterbi algorithm
2. Sequential decoding – Stack and Fano algorithm

6.1 CONVOLUTIONAL CODE DECODING STRATEGIES:

6.1.1 MAXIMUM LIKELY HOOD DECODING:
If all input messages are equally likely, a decoder that achieves the minimum probability
of error is one that compares the conditional probabilities, also called the likelihood
functions P(Z|U(m)), where Z is the received sequence and U(m) is one of the possible
transmitted sequences, and chooses the maximum. The decoder chooses U(m’) if

The maximum likelihood concept is a fundamental development of decision theory. Since

the channel coding employed in the project is convolutional, it is evident that this approach
is to be applied to the decoding of convolutional codes. As convolutional codes have
memory, this application of maximum likelihood approach would be in the context of
selecting the most likely sequence. Therefore, in the maximum likelihood context, one can
say that the decoder chooses a particular U(m’) as the transmitted sequence if the likelihood

38
P(Z|U(m’)) is greater than the likelihoods of all other possible transmitted sequences. For a
convolutional code of rate 1/n, we can therefore express the likelihood as:

Such an optimal decoder, which minimizes the error probability, is known as maximum
likelihood decoder. The maximum likelihood decoding has been done using the trellis
representation of the codes because with such a representation, it is possible to configure a
decoder that can discard the paths that could not possibly be candidates for the maximum
likelihood sequence.

6.1.2 FEATURES OF MAXIMUM LIKELYHOOD DECODING:

• When the encoded information is transmitted over the channel, it is distorted.

• The convolutional decoder regenerated the information by estimating the most likely path
of state transition in the trellis.

• Maximum likelihood decoding means the decoder searches all the possible paths in the
trellis and compares the metrics between each path and the input sequence. The path with
the minimum metric is selected as the output. So maximum likelihood decoder is the
optimum decoder.

• In general, a convolutional code (n, k, K) has 2(K-1)k possible states. At a sampling

instant, there are 2k merging paths for each node and the one with the minimum distance is
selected and called surviving path. At each instant, there are 2(K-1)k surviving paths are
stored in the memory.

• When all the input sequence are processed, the decoder will select a path with the
minimum metric and output it as the decoding result.

• In the real systems, the input sequence is very long. It is showed that it is long enough
to make the trellis depth L> (5*K) and decode only the newest message bit within the
Viterbi trellis.

39
6.1.3 VITERBI ALGORITHM:
The Viterbi algorithm essentially performs maximum likelihood decoding; however, it
reduces the computational load by taking advantage of the special structure of the code
trellis. The advantage of the Viterbi decoder is that the complexity of such a decoder is not
a function of the number of the symbols in the codeword sequence. The algorithm involves
calculating a measure of similarity or difference, between the received signal, at time ti, and
all the trellis paths entering each state at time ti. The Viterbi algorithm removes from
consideration all those trellis paths that could not possibly be candidates for the maximum
likelihood choice. When two paths are entering the same state at some time ti, then the one
having the better metric is chosen; this path is then called the surviving path. The selection
of surviving path is then repeated for all the states and thereafter the entire process is
repeated for all the subsequent time instants. The decoder continues to advance deeper into
the trellis, making decisions by eliminating the least likely paths. The early rejection of the
least likely paths is significant in reducing the decoder complexity. The goal of selecting
the optimum path may be equivalently expressed as choosing the codeword with maximum
likelihood metric, or as choosing the codeword with minimum distance metric.

1.Calculation Of Branch Matrics: The branch metric at time instant j for path, is defined
as the joint probability of the received n-bit symbol conditioned on the estimated
transmitted n-bit symbol :

2. Calculation of path metrics: The path metric for path , at time instant J is the sum of
the branch metrics along this path.

40
3. Information sequence update: At each instant, there are 2k merging paths for every
node. The decoder selects the one with the largest metric as the survivor.

4. Outputting the decoded sequence: At instant J, the (J-L)th information symbol is

output from the memory with the largest metric.

ALGORITHM:
Let m(s,t) represent the matrix of state s at time t.
1. Initially, all state metrics are zeros, i.e. m(0,0) = m(1,0) = m(2,0) = m(3,0) = 0.

2. For every state, there are two entering branches, called upper branch and lower branch
respectively. Variables M_upper (s, t) and M_lower (s, t) are used to stand for Hamming
distance between the current input information bits and expected information bits which
cause the state transition to s at time t.

3. Compare M_upper (s, t) + m(s*, t-1) and M_lower (s, t) + m(s*, t-1), where s is the state
at time t, and s* is the pervious state at time (t-1) for a given branch. Choose the branch
with the small value as the surviving branch entering the state at time t and let m(s, t)= this
value. That is: If M_upper (s, t) + m(s*, t-1) < M_lower (s, t) + m(s*, t-1)
The upper branch is the surviving branch and m(s,t)= M_upper (s, t) + m(s*, t-1).
Otherwise, the lower branch is the surviving branch and m(s,t)= M_lower (s, t) + m(s*, t-
1) Repeat above steps until all the input data are calculated at time T.

4. Compare m(0, T) m(1,T) m(2, T) m(3, T), choose the minimum one and trace back the
path from this state. The output data can be generated corresponding to the path.

6.1.4 SEQUENTIAL DECODING:

A sequential decoder works by generating hypotheses about the transmitted codeword
sequence; it computes a metric between these hypotheses and the received signal. It goes
forward as long as the metric indicates that its choices are likely; otherwise, it goes
backward, changing hypotheses until, through a systematic trial-and-error search, it finds
likely hypothesis. Sequential decoders can be implemented to work with hard or soft
decisions, but soft decisions are usually avoided because they greatly increase the amount

41
of the required storage and the complexity of the computations. The two most prominent
decoding algorithms that implement sequential decoding are Stack algorithm, Fano
algorithm. All the decoding strategies discussed above would be able to recover the data
from the channel only if it has been corrupted by noise source that is statistically
independent, that is additive noise, or the case when the successive noise bits are
independent.
Unfortunately, the Viterbi algorithm performs poorly when it is presented with bit errors
that are all bunched together in the stream, and because the sub carriers are subject to flat
fading bit errors usually do occur in groups when a subcarrier is in a deep fade. To protect
against this, time interleaving and frequency interleaving is used.

SUMMARY:
We saw the different decoding techniques. In our project we use viterbi decoding
technique.

42
 CHAPTER 7

ADVANTAGES
&
DISADVANTAGES

43
 ADVANTAGES, DISADVANTAGES

7.1 ADVANTAGES OF OFDM:

OFDM possesses some inherent advantages for Wireless Communications. This section
glances on few of the most important reasons on why OFDM is becoming more popular
in the Wireless Industry today.

 HIGH SPECTRAL EFFICIENCY:

High spectral efficiency in OFDM is achieved mainly due to the use of orthogonal sub-
carriers that would allow the sub-carriers spectra to overlap. Also for a given overall data
rate,increasing the number of carriers reduces the data rate that each individual carrier must
convey, and hence lengthens the symbol period. This means that the intersymbol
interference affects a smaller percentage of each symbol as the number of carrier and hence
the nuber of symbol period increases.

 MULTIPATH DELAY SPREAD TOLERANCE:

The increase in the symbol time of the OFDM symbol by N times (N being the number of
sub-carriers), leads to a corresponding increase in the effectiveness of OFDM against the
ISI caused due to multi-path delay spread. Further, using the cyclic extension process and
proper design, one can completely eliminate ISI from the system.

 EFFECTIVENESS AGAINST CHANNEL DISTORTION:

In addition to delay variations in the channel, the lack of amplitude flatness in the
frequency response of the channel also causes ISI in digital communication systems. A
typical example would be the twister-pair used in telephone lines. These transmission lines
are used to handle voice calls and have a poor frequency response when it comes to high
frequency transmission. In systems that use single-carrier transmission, an equalizer might
be required to mitigate the effect of channel distortion. The complexity of the equalizer
depends upon the severity of the channel distortion and there are usually issues such as
equalizer non-linearities and error propagation etc that cause additional trouble. In OFDM
systems on the other hand, since the bandwidth of each sub-carrier is very small, the
amplitude response over this narrow bandwidth will be basically flat (of course, one can
safely assume that the phase response will be linear over this narrow bandwidth). Even in

44
the case of extreme amplitude distortion, an equalizer of very simple structure will be
enough to correct the distortion in each sub-carrier.

 THROUGHPUT MAXIMISATION:
The use of sub-carrier modulation improves the flexibility of OFDM to channel fading and
distortion makes it possible for the system to transmit at maximum possible capacity using
a technique called channel loading. Suppose the transmission channel has a fading notch in
a certain frequency range corresponding to a certain sub-carrier. If we can detect the
presence of this notch by using channel estimation schemes and assuming that the notch
doesn’t vary fast enough compared to the symbol duration of the OFDM symbol, it can be
possible to change (scale down/up) the modulation and coding schemes for this particular
sub-carrier (i.e, increase their robustness against noise), so that capacity as a whole is
maximized over all the sub-carriers. However, this requires the data from channel-
estimation algorithms. In the case of single-carrier systems, nothing can be done against
such fading notches. They must somehow survive the distortion using error correction
coding or equalizers.

 ROBUSTNESS AGAINST IMPULSE NOISE:

Impulse noise is usually a burst of interference caused usually caused in channels such as
the return path HFC (Hybrid-Fiber-Coaxial), twisted-pair and wireless channels affected by
atmospheric phenomena such as lightning etc. It is common for the length of the
interference waveform to exceed the symbol duration of a typical digital communication
system. For example, in a 10 MBPS system, the symbol duration is 0.1 s, and a impulse
noise waveform, lasting for a couple of micro-seconds can cause a burst of errors that
cannot be corrected using normal error-correction coding. Usually complicated Reed-
Solomon codes in conjunction with huge interleaves are used to correct this problem.
OFDM systems are inherently robust against impulse noise, since the symbol duration of an
OFDM signal is much larger than that of the corresponding single-carrier system and thus,
it is less likely that impulse noise might cause (even single) symbol errors. Thus,
complicated error-control coding and interleaving schemes for handling burst-type errors
are not really required for OFDM Systems simplifying the transceiver design.

45
7.2 DISADVANTAGES OF OFDM:
There are two main issues associated with OFDM. They are Peak to Average Power ratio
and time and frequency synchronization.

 PEAK to AVERAGE POWER RATIO:

OFDM exhibits a high peak-to-average ratio. In other words, there is a problem of extreme
amplitude excursions of the transmitted signal. The OFDM signal is basically a sum of N
complex random variables, each of which can be considered as a complex modulated signal
at different frequencies. In some cases, all the signal components can add up in phase and
produce a large output and in some cases, they may cancel each other producing zero
output.
Distortion in the transmitter can result in spectral spreading, which can cause interference
to neighboring systems in RF frequency. In addition, the power amplifier may saturate and
clip the signal. The clipping of the signal will reduce SNR and increase the BER. Also it
results in bandwidth leakage.
In order to avoid clipping of the transmitted waveform, the power-amplifier at the
transmitter front-end must have a wide linear range to include the peaks in the transmitted
waveform. Another way involves pre-distorting the signal before the power amplifier in
such a way as to cancel the distortion caused by the power amplifier.

 FREQUENCY & PHASE ERROR SENSITIVE:

OFDM is very sensitive to frequency and phase errors between the transmitter and receiver.
The main sources of these errors are frequency stability problems, any frequency offset
errors between the transmitter and receiver or phase noise of of the transmitter. This
problem can be mostly overcome by synchronizing the clocks between the transmitter and
receiver, by designing the system appropriately, or by reducing the number of carriers used.

46
 CHAPTER 8

OFDM
OVER
FDMA, TDMA, CDMA

OFDM OVER FDMA,TDMA,CDMA

47
In typical FDMA system up to 50% the total spectrum is wasted due to the extra spacing
between channels. This problem becomes worse as the channel BW becomes narrower, and
the frequency band increases.

TDMA partly overcomes this problem by using wider bandwidth channels, which are used
by several users. However there are two main problems with TDMA. There is an overhead
associated with the change over between users due to time slotting on the channel which
limits the number of users that can be sent efficiently in each channel. in addition, the
symbol rate of each channel is high resulting in problems with multipath delay spread.

OFDM overcomes most of the problems with both FDMA and TDMA. Because of
orthogonality and no need for users to be time multiplexed as in TDMA.

Most third generation mobile phone systems are proposing to use Code Division Multiple
Access (CDMA) as their modulation technique. It was found that OFDM performs
extremely well compared with CDMA, providing a very high tolerance to multipath delay
spread, peak power clipping and channel noise. In addition to this it provides a high
spectral efficiency as the carrier power and modulation scheme can be individually
controlled for each carrier.

48
 CHAPTER 9

APPLICATIONS
OF
OFDM

APPLICATIONS OF OFDM

49
A lot of applications that use OFDM technology have spawned over the last few years.
They include:

 DIGITAL AUDIO BROADCASTING (DAB)

DAB is an European standard for digital broadcasting that is intended to replace the current
analog technologies such as AM and FM. It was standardized by the European
Telecommunications Institute (ETSI) in 1995.

DAB is designed to be a single frequency network, in which the user receives same signals
from several different transmitters. This greatly enhances spectral efficiency. Even though
there is a delay in the reception of signals from different transmitters, this situation can be
considered as a multi-path situation and can be easily handled by selecting the guard
interval properly.

 DIGITAL VIDEO BROADCASTING (DVB)

Digital Video Broadcasting (DVB) is a standard for broadcasting Digital Television over
satellites, cables and through terrestrial (wireless) transmission. Many digital video
broadcasting services, including but not restricted to digital television, are currently
available or being developed. Two main factors are behind the decision to use OFDM for
DVB-T and satellite broadcasting services. One is its robustness against narrow band
interference (with coding and interleaving) due to existing analog systems. The other
reason OFDM is preferred is its performance against multi-path, which is very common for
terrestrial broadcast channels.

WIRELESS LANS
Wireless LANs are one of the most important applications of OFDM. A lot of standards
have been proposed for Wireless LANs during the past decade, most of then based on
spread-spectrum schemes. In July 1998, IEEE Wireless LAN standardization group IEEE
802.11 standardized a scheme based on OFDM operating in the 5-GHz band. It is
interesting to note that this standard is one of the first packet-based one to use OFDM.

50
 CHAPTER 10

FUTURE SCOPE

51
 FUTURE SCOPE
OFDM has been chosen for several current and future communication systems all over the
world. Modern digital signal processing technique now make it possible to use this
modulation system to improve the reliability of the communication link. OFDM is well-
suited for systems in which the channel characteristics make it difficult to maintain
adequate communications link performance.

Asyncronous digital subscriber line(ADSL) provides a method of delivering high speed

data over the phone line. The system uses Ofdm techniques, calling their variation discrete
multi-tone(DMT). DMT includes features for allowing the removal of subcarriers and for
adjusting modulation format on a per carrier basis to best suit the transmission channel
characteristics.

European digital television standard makes use of OFDM. Discussions are going in the
U.S. to look at a similar system and Japan is close to adopting a similar standard for their
future digital TV broadcast system.

The next generation of radio broadcast may also make use of OFDM techniques. Various
High-speed wireless networking standards in the 5 GHz frequency region employ OFDM
modulation. Various data rates from 6 to 54 Mbps are possible. OFDM works well in home
and office environments for handling wall reflections and movements within the structure.

52
 CHAPTER 11

RESULTS

53
 RESULTS

Time domain signal

Frequency domain signal

54
OFDM WITH CONVOLUTION CODE

OFDM WITH HAMMING CODE

55
OFDM WITH & WITHOUT CONVOLUTION CODE

CONVOLUTION WITH DIFFERENT CODE RATES

56
OFDM WITH INTERLEAVER

OFDM WITH EQUALIZER

57
 CHAPTER 12

CONCLUSION

58
 CONCLUSION

In this project, performance of the OFDM system has been studied on the basis of BER. It
was found that the performance of the system is improved with the use of different channel
coding, interleaving and equalizing techniques. As seen from the graphs, a worse BER
performance is observed with no coding and best performance with coding and
equalization.

OFDM provides the best of the benefits of FDMA, TDMA and CDMA while avoiding the
limitations of each. The ability to define the signal in the frequency domain, in software on
VLSI processors, and to generate the signal using IFFT is the key to its current popularity.
The advances in VLSI technology have made it possible to implement efficiently an IFFT
and FFT block in hardware.

Multi –user OFDM allow for highly flexible communications thus can be made to adapt to
radio channel condition resulting in high spectral density & reliability.

Thus, OFDM techniques are quickly becoming a popular method for advanced
communication networks.

59
 Abstract

The basic idea of OFDM is to divide the available spectrum into many narrow bands, low
data rate carriers. To obtain high spectral efficiency, the frequency response of the sub-
carriers are overlapping and orthogonal, hence the name OFDM.

Each narrow band sub-carrier can be modulated using various modulation formats where
BPSK,QPSK and QAM are commonly used. OFDM modulation and demodulation are
typically implemented using digital filter bank generally using the FFT. After each OFDM
symbol cyclic prefix is inserted this removes ISI & ICI.

The benefits of using OFDM are many, including high spectral efficiency, resistance
against multi-path interference, and ease of filtering out noise.

Thus OFDM has emerged as a very promising technology to provide high data rate
transmission over base-band channels.

60