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IMPLEMENTATION OF OFDM SYSTEM
CHAPTER 1
INTRODUCTION
Orthogonal frequency division multiplexing (OFDM) is one of the multi-carrier

modulation techniques that transmit signals through multiple carriers. These carriers
(subcarriers) have different frequencies and they are orthogonal to each other. Orthogonal
frequency division multiplexing techniques have been applied in both wired and wireless
communications, such as the asymmetric digital subscriber line (ADSL) and the IEEE
802.11 standard. It is well known that Chang proposed the original OFDM principles in
1966, and successfully achieved a patent in January of 1970. Later on, Salzburg analyzed
the OFDM performance and observed that the crosstalk was the severe problem in this
system. Although each subcarrier in the principal OFDM systems overlapped with the
neighborhood subcarriers. The orthogonality can still be preserved through the staggered
QAM (SQAM) technique. However, the difficulty will emerge when a large number of
subcarriers are required. In some early OFDM applications, the number of subcarriers can
be chosen up to 34. Such 34 symbols will be appended with redundancy of a guard time
interval to eliminate inter symbol interference (ISI).
However, should more subcarriers be required, the modulation, synchronization,

and coherent demodulation would induce a very complicated OFDM requiring additional
hardware cost. In 1971, Weinstein and Ebert proposed a modified OFDM system in
which the discrete Fourier Transform (DFT) was applied to generate the orthogonal
subcarriers waveforms. Their scheme reduced the implementation complexity
significantly, by making use of the IDFT modules and the digital-to-analog converters. In
their proposed model, baseband signals were modulated by the inverse DFT (IDFT) in
the transmitter and then demodulated by DFT in the receiver. Therefore, all the
subcarriers were overlapped with others in the frequency domain, while the DFT
modulation still assures their orthogonality, as shown in Fig. Moreover, the windowing
technique was introduced in this paper to attack the inter-symbol interference (ISI) and
BAPATLA WOMEN’S ENGINEERING COLLEGE 1

inter-carrier interference (ISI) problems. Thanks to these evolutions, modern low-cost

OFDM systems have become plausible.
Fig: 1.1 Frequency spectrum of OFDM subcarrier signals
Cyclic prefix (CP) or cyclic extension was first introduced by Peled and Ruiz in
1980 for OFDM systems. In their scheme, conventional null guard interval is substituted
by cyclic for fully-loaded OFDM modulation. As a result, the orthogonality among the
subcarriers was guaranteed. With the trade-off of the transmitting energy efficiency, this
new scheme can result in a phenomenal ICI reduction. Hence it has been adopted by the
current IEEE standards.
In 1980, Hirosaki introduced an equalization algorithm to suppress both ISI and,

which may have resulted from a channel distortion, synchronization error, or phase error.
In the meantime, Hirosaki also applied QAM modulation, pilot tone, and trellis coding
techniques in his high-speed OFDM system, which operated in voice-band spectrum. In
1985, Cimini introduced a pilot based method to reduce the interference emanating from
the multipath and co-channels. In 1989, Kalet suggested a subcarrier-selective allocating
scheme. He allocated more data through transmission of “good” subcarriers near the
center of the transmission frequency band; these subcarriers will suffer less channel

distortion. In the 1990s, OFDM systems have been exploited for high data rate
communications. In the IEEE 802.11 standard, the carrier frequency can go up as high as
2.4 GHz or 5 GHz. Researchers tend to pursue OFDM operating at even much higher
frequencies nowadays. For example, the IEEE 802.16 standard proposes yet high carrier
frequencies ranging from 10 GHz to 60 GHz.
1.1MOTIVATION
The demand for future higher data rate communications always provides the
impetus for this research. It is obvious that a parallel system is capable of carrying more
information than a cascade system, simply because it uses a variety of frequency bands.
However, the significant advantage of OFDM is that it is robust in frequency-selective
channels, which result from either multipath fading or other communication
interferences. The interference problem is severe especially for some systems working in
the Unlicensed National Information Infrastructure (U-NII) operating frequency range,
such as the IEEE 802.11a standard. Under any of these conditions, the channel has non
uniform power gains, as well as nonlinear phases across frequencies. An example is
provided in Fig. 1.2
Fig: 1.2 Magnitude response of a fading channel
In order to deal with frequency-selective fading’s, the transmitted OFDM signals

are divided into many sub-channels so that those sub-channels can be considered

frequency- flat approximately as the number of the sub-channel N is large enough. Hence
the OFDM signals will suffer channel distortion less than the conventional modulated
signals. Under OFDM modulation, the symbol duration becomes N times longer. For
example, if the input data rate is 20Mbps, then, the symbol duration is 50 ns however, in
an OFDM system with 128 subcarriers, the symbol duration could become 6.4 us. If these
two kinds of symbols are modulated and transmitted through a channel with a particular
rms value say, rms = 60 ns, it is clear that the system with longer symbol duration would
perform better. In practice, the DVBT standard suggests to use 2,048 subcarriers, or
8,192 sub carriers. In these cases, the symbol duration can be even increased by several
thousand times of ofdm.
1.2 RELATED TECHNOLOGIES
Single Input and Single Output (SISO) OFDM is the one in which only one
transmitting antenna and one receiving antenna are used for transmission. If multiple
transmitting and receiving antennas are used for transmission then it is called Multiple
Input Multiple Output (MIMO) OFDM. MIMO-OFDM uses space time codes to ensure
that the signals transmitted over the different antennas are orthogonal to each other.
Orthogonal Frequency Division Multiple Access (OFDMA) is the combination of OFDM
with Time Division Multiple Access (TDMA), in which number of time slots are allotted
for different users.
Multi-Carrier Code Division Multiple Access (MC-CDMA), is multi user version

of OFDM with the combination of CDMA. Coded OFDM (COFDM) is a form of OFDM
in which error correction coding is incorporated into the OFDM signal. OFDM with
CDMA will be the future scope for the need of higher data rates for transferring high
definition videos and audios, online gaming, entertainment etc.
The OFDM techniques had been applied for ANDEFT and KINEPLEX, since
1960s. After the IFFT/FFT technique was introduced, the implementation of OFDM

became more convenient. Generally speaking, the OFDM applications may be divided
into two categories-wired and wireless technologies. In wired systems such as
Asymmetric Digital Subscriber Line (ADSL) and high speed DSL, OFDM modulation
may also be referred as Discrete Multi tone Modulation (DMT). In addition, wireless
OFDM applications may be shown in numerous standards such as IEEE 802.11 and
Hiper LAN.
OFDM was also applied for the development of Digital Video Broadcasting
(DVB) in Europe, which was widely used in Europe and Australia. In the DVB standards,
the number of subcarriers can be more than 8,000, and the data rate could go up as high
as 15Mbps. At present, many people still work to modify the IEEE 802.16 standard,
which may result in an even higher data rate up to 100Mbps.

CHAPTER 2
OFDM
In this chapter, principal and modern OFDM systems will be discussed. The
principal model, as well as mathematical formulae, will be addressed to demonstrate the
advantages of the OFDM technique. In addition, special problems will be explored,
which might occur in OFDM systems. Finally an overview of the Peak-to-average-power
Ratio (PAPR) problem will be provided in this chapter.
2.1 PRINCIPLE OF OFDM MODEL

2.1.1 Over view
Since the original OFDM model was proposed in the 1960s the core structure of
OFDM has hardly changed. The key idea of OFDM is that a single user would make use
of all orthogonal subcarrier in divided frequency bands. Therefore, the data rate can be
increased significantly. Since the bandwidth is divided into several narrower sub
channels, each sub channel requires a longer symbol period. Channel bandwidth is
divided into multiple sub channels to reduce ISI and frequency-selective fading.
Multicarrier transmission data is transmitted in parallel on multiple carriers that overlap
in frequency and also are orthogonal to each other. Subcarriers are orthogonal each other
in frequency domain.
Therefore OFDM systems can overcome the inter symbol interference (ISI)
problem. As a consequence, the OFDM system can result in lower bit error rates but
higher data rates than conventional communication systems. Nevertheless, OFDM
technique has certain drawbacks such as the increased system complexity, which is
associated with the generation of orthogonal subcarriers, and other new problems, which
might not occur in other modulation schemes. Such new problems include the peak to
average power ratio and inter-symbol interference.

2.1.2 Ofdm modulation
Fig: 2.1 serial-to-parallel conversion
As depicted the input data stream is converted into N parallel data streams
through a serial-to-parallel port. The duration of the data is elongated by N times. Serial-
to-parallel conversion is depicted in Fig 2.1. 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.

Fig 2.2: N - Times serial – parallel conversion
2.1.3 Orthogonal Subcarriers
When the parallel symbol streams are generated, each stream would be modulated and
carried at different center frequencies as the traditional FDM scheme. the effect of increasing
the subcarriers was investigated with data rate of 1 Mbps transmitted over the frequency
selective multipath channel at different mobile speeds. The OFDM-4QAM signalling
scheme which provides way of parallel transmission are compared to analyze the BER
performance of the designed OFDM system. The results obtained showed that as the
number of subcarriers increases, the BER also increases accordingly, which indicates
poor performance.

Fig.2.3: An example of subcarrier waveforms.
2.1.4 Band pass Signaling
After the modulation by orthogonal sub carriers, all N subcarrier waveforms were
added together to be up-converted to the pass-band. This resulting signal waveform will
transmitted with a carrier frequency at 2.4G Hz, 5G Hz, 11G Hz or 60 G Hz. Then the
band-pass OFDM signal waveform would be sent to power amplifier and antennas.
Thus, transmitted OFDM signal x(t) can be expressed as:
N−1
exp ⁡( j2 π ( f c +n ∆ f ) t)
x(t) = ∑ s(n) (2.1.4a)
n =0 √T

Where the s(n) represents the input data stream, T the symbol duration and the
carrier frequency. Traveling through a wireless channel, the distorted x(t) results in r(t),
as shown in Fig.
2.1.5 Ofdm Demodulation
Fig 2.4: OFDM Demodulation
At the receiver, the received signal is down-converted to form a base-band signal.

Then, low-pass filters and de-subcarriers are applied to separate subcarrier waveforms.
Orthogonality of sub-carriers will ensure that only the targeted subcarrier waveform will
be preserved in each sub-band. Ideally, the final detected symbols will be identical to
those transmitted in figure.
2.2 MODERN OFDM MODEL
It was a complicated process to achieve N subcarrier oscillators, especially when

The N was large. Therefore, the principal OFDM model could not been widely used.

Thanks to IDFT/DFT-based algorithms introduced in 1980. OFDM has become a

wireless modulation technique since after. The modern OFDM model is depicted.
Fig 2.5: OFDM model
2.2.1 IDFT and Orthogonality
After the parallel symbol streams are generated, IDFT operator substitutes the
aforementioned local subcarrier oscillators. IDFT operation, F (n) f (k), can be
Written as
N −1
1
F (k) =
√N
∑ F (n)exp ⁡( j2Nπ kn ¿ )¿ (2.2.1a)
n=0
The orthogonality of discrete-time Fourier bases can be described as follows:

N−1
2 π kn
∑ cos ( 2 πNkn ) ×cos( N
¿=δ (n−m) (2.2.2b)
k=0
Clearly, the output sequences of the IDFT are orthogonal to each other. With the absence
of local oscillators, the OFDM system complexity has been greatly reduced.
2.3 GUARD INTERVAL AND CYCLIC PREFIX

2.3.1 Introduction to cyclic prefix
In wireless channel, due to multipath propagation and fading the broadening of

symbol occurs. The broadening of the symbols causes the interference of one symbol
with the next symbol. This is called as inter symbol interference (ISI). Due to ISI the start
of the symbol will be lost.
Due to Doppler Effect, the frequency of subcarriers changes, which will cause the
loss of orthogonality among the subcarriers. As these subcarriers are closely spaced and if
the orthogonality is lost, the subcarriers will interfere with each other. This is called as
inter carrier interference. Due to the information in the subcarriers is lost/ corrupted.
To avoid the inter symbol interference (ISI) and inter carrier interference, cyclic
extensions (cyclic prefix, cyclic postfix) are used. Normally cyclic prefix is used. Cyclic
prefix is the addition of a part of the tail of OFDM symbol at the start of the OFDM
symbol. The inter symbol interference (ISI) due to symbol spreading is shown in Fig. 2.5
and cyclic prefix extension used to avoid the inter symbol interference (ISI) is shown in
Fig The cyclic prefix acts as a buffer region or guard interval to protect the OFDM
signals from intersymbol interference. This can be an issue in some circumstances even
with the much lower data rates that are transmitted in the multicarrier OFDM signal.The
basic concept behind the OFDM cyclic prefix is quite straightforward.The cyclic prefix is
created so that each OFDM symbol is preceded by a copy of the end part of that same

symbol.Different OFDM cyclic prefix lengths are available in various systems. For
example within LTE a normal length and an extended length are available and after
Release 8 a third extended length is also included, although not normally used.
Transmitted symbol
length
Inter Symbol
Received symbol Interference
length
Fig.2.6: Inter Symbol Interference due to Symbol Broadening

Guard
band
Transmitted symbol
length
Received symbol
Cyclic
length
prefix
Fig.2.7: Cyclic Prefix is inserted to avoid Inter Symbol Interference
2.3.2 Introduction to Pilot
In wireless medium, the channel characteristics are both time varying and
frequency varying. If the OFDM signal is transmitted into this channel, it undergoes
channel effects. To recover the original transmitted data from the received signal, the
channel effects have to be equalized. For this, first the channel characteristics have to be
estimated. Using this estimated channel characteristics, the channel equalization is done.
There are number of methods for the channel estimation, mainly they are divided into
two groups.
i. Non data aided channel estimation

ii. Data aided channel estimation
In non data aided channel estimation, no additional data is transmitted along the
user data, the channel estimation is done using the channel statistics and some of the
transmitted signal properties. For this channel estimation long data records are required,

hence it is not applicable to fast fading channel. For fast fading channels, data aided
channel estimation techniques are used. In data aided channel estimation techniques,
training symbols or pilot data bits are inserted into the user data bits which are known
prior to the receiver. As the pilot bits are known prior to receiver, it is easy to estimate the
channel characteristics.
In figure the block denoted as “Add Cyclic Prefix” actually should be called the
“Add Guard Interval” in general. The cyclic prefix (CP) is the most common guard
interval (GI). The GI is introduced initially to eliminate the inter block interference (IBI).
Since one block of input data symbols are associated with a single transmitted waveform
in an OFDM system, most people refer IBI as ISI. Figure demonstrates how to use the GI
to eliminate the ISI. However, multipath fading channel models are concerned in most
situations. Therefore, many time-delayed visions of the transmitted waveform might be
found at the receiver. Without GI, these waveforms would interfere with each other, just
as demonstrated in the top half of figure Nevertheless, in those cases where the GI was
employed, the portions of waveforms received in the GI duration would be totally
discarded, as shown in the bottom half of figure Thus, the ISI could be completely
eliminated accordingly. It is noted that the GI duration must be larger than the maximum
channel delay time. Otherwise, it could not entirely remove the ISI.

Fig 2.8: GI Eliminates ISI
There are several options for GI. One choice of GI is zero padding. In this
scheme, no waveform is transmitted in the GI duration. However, the zero-padded
waveform would destroy the orthogonality of subcarriers and results in inter symbol
interference (IsI). The cyclic prefix (CP) is a good substitute of the zero-padding GI. In
the CP scheme, the GI is a copy of the partial waveform. Based on the fact that the
Fourier bases are periodic functions, the orthogonality of subcarriers can be preserved
consequently. As depicted in figure, an end-portion of waveform is copied and inserted
prior to the beginning of waveform. The time duration of CP, denoted as G T in the figure,
is often choosen according to the following:

Fig.2.9: cyclic prefix
For example, in the IEEE 802.11a standard, k 2 is chosen; in Europe DVB
(Digital Video Broadcasting) standards, k 1,2,...6 can be employed. As aforementioned,
k should also depend on the maximum delay time of the channel. When CP is applied
instead of zero-padding GI, both ICI and ISI are eliminated. D. A Practical OFDM
System- HiperLAN/2 Even though there are many applications of the OFDM technique,
the corresponding parameters vary from one to another and depend on the practical
purposes. Fig 2.8 illustrates the simplified version of a practical OFDM system--the
HiperLAN/2 system. As a wireless LAN model, the HiperLAN/2 standard can provide
services at 54 Mbps data-rate. The European Telecommunication Standards Institute
(ETSI) and Broadband Radio Access Network (BRAN) originally proposed this standard.
At present, it has been widely applied in airports, homes, and universities in Europe,
Australia and Japan. In this standard, many techniques such as QAM, convolution
coding, puncture, and interleaver might be used. At the receiver, there are de-interleaver,

zero-inserting, and Viterbi decoder components correspondingly. The OFDM modulation

in the HiperLAN/2 system is similar the principal one, except that four pilot bits were us
ed for the time and frequency synchronization, and CP is inserted prior to the OFDM
information symbols, as depicted in figure. As a matter of fact, the HiperLAN/2 standard
is nearly the same as the IEEE 802.11a standard at the physical layer; both standards use
64 subcarriers with 16 cyclic prefix bits per block.
Fig.2.10: simulink model of OFDM.

Choosing communication hardware components with extremely broad linear
range could mitigate this power, but that would also badly decrease system efficiency, or
introduce excess noise. Those drawbacks can be explained as follows. a large dynamic
range of the D/A and A/D converters will be required; otherwise, the peak values could

be clipped, which would result in a signal distortion. On the other hand, if A/D and D/A
converters with large working ranges are chosen, the quantization noise will increase and
the system performance will degrade.
A large working range is required to ensure that the nonlinear distortion would
not be introduced. As a result, power efficiency is decreased significantly due to PAPR
problem. For example, the maximum power efficiency of a Class B power amplifier
drops from 78.5% to 4.6%, when the PAPR increases 0dB to 17dB, as stated in the IEEE
802.11a standard. Power efficiency is pivotal in mobile communications such as laptop
and PDA over the wired communications. Since the PAPR problem is not the focus of
this thesis, only an overview of existing solutions for is provided here. In fact, there are
many approaches, such as coding, partial transmit sequences, selective mapping, clipping
random phase updating algorithm, and sub-block phase weighting.
CHAPTER 3
FOURIER TRANSFORM AND BLOCKS

The Fourier transform allows us to relate events in time domain to events in

frequency domain. There are several version of the Fourier transform, and the choice of
which one to use depends on the particular circumstances of the work. The conventional
transform relates to continuous signals which are not limited to in either time or
frequency domains. However, signal processing is made easier if the signals are sampled.
Sampling of signals with an infinite spectrum leads to aliasing, and the processing of
signals which are not time limited can lead to problems with storage space. To avoid this,
the majority of signal processing uses a version of the discrete Fourier transform (DFT).
The DFT is a variant on the normal transform in which the signals are sampled in both
time and the frequency domains. By definition, the time waveform must repeat
continually, and this leads to a frequency spectrum that repeats continually in the
frequency domain.
The fast Fourier transform (FFT) is merely a rapid mathematical method for
computer applications of DFT. It is the availability of this technique, and the technology
that allows it to be implemented on integrated circuits at a reasonable price, that has
permitted OFDM to be developed as far as it has. The process of transforming from the
time domain representation to the frequency domain representation uses the Fourier
transform itself, whereas the reverse process uses the inverse Fourier transform.
3.1 THE USE OF FFT IN OFDM
The main reason that the OFDM technique has taken a long time to become a
prominence has been practical. It has been difficult to generate such a signal, and even
harder to receive and demodulate the signal. The hardware solution, which makes use of
multiple modulators and demodulators, was somewhat impractical for use in the civil
systems. The ability to define the signal in the frequency domain, in software on VLSI
processors, and to generate the signal using the inverse Fourier transform is the key to its
current popularity. The use of the reverse process in the receiver is essential if cheap and
reliable receivers are to be readily available. Although the original proposals were made a
long time ago, it has taken some time for technology to catch up. At the transmitter, the

signal is defined in the frequency domain. It is a sampled digital signal, and it is defined
such that the discrete Fourier spectrum exists only at discrete frequencies. Each OFDM
carrier corresponds to one element of this discrete Fourier spectrum. The amplitudes and
phases of the carriers depend on the data to be transmitted. The data transitions are
synchronized at the carriers, and can be processed together, symbol by symbol.
The frequency division multiplexed signal illustrates typical FFT-based OFDM

system. The incoming serial data is first converted form serial to parallel and grouped
into x bits each to form a complex number. The number x determines the signal
constellation of the corresponding subcarrier, such as 16 QAM or 32QAM. The complex
numbers are modulated in the baseband by the inverse FFT (IFFT) and converted back to
serial data for transmission. A guard interval is inserted between symbols to avoid
intersymbol interference (ISI) caused by multipath distortion. The discrete symbols are
converted to analog and low-pass filtered for RF up conversion. The receiver performs
the inverse process of the transmitter. One-tap equalizer is used to correct channel
distortion.
Fig 3.1: Example of the power spectral density of the OFDM signal with a guard
interval D = TS/4 (number of carriers N=32)]
3.2 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 the
transmitted in parallel by assigning each data word to one carrier in the transmission.
3.2.1 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.
3.3 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.
3.3.1 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.

3.3.2 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.
3.3.3 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.
3.4 OFDM SIMULATION PARAMETERS
The configuration used for most of the simulations performed on the OFDM
signal. An 800-carrier system was used, as it would allow for up to 100 users if each were
allocated 8 carriers. The aim was that each user has multiple carriers so that if several
carriers are lost due to frequency selective fading that the remaining carriers will allow
the lost data to be recovered using forward error correction. For this reason any less then
8 carriers per user would make this method unusable. Thus 400 carriers or less was
considered too small. However more carriers were not used due to the sensitivity of
OFDM to frequency stability errors.
The greater the number of carriers a system uses, the greater it required frequency
stability. For most of the simulations the signals generated were not scaled to any

particular sample rate, thus can be considered to be frequency normalized. Three carrier
modulation methods were tested to compare their performances. This was to show a
tradeoff between system capacity and system robustness. DBPSK gives 1 b/Hz spectral
efficiency and is the most durable method, however system capacity can be Increased
using DQPSK (2 b/Hz) and D16PSK (4 b/Hz) but at the cost of a higher BER. The
modulation method used is shown as BPSK, QPSK, and 16PSK on all of the simulation
plots, because the differential encoding was considered to be an integral part of any
transmission.
3.4.1 Ofdm frame conversion
The Frame Conversion block passes the input through to the output and sets the
output sampling mode to the value of the Sampling mode of output signal parameter,
which can be either Frame-based or Sample-based. The output sampling mode can also
be inherited from the signal at the Ref (reference) input port, which you make visible by
selecting the Inherit output sampling mode from <Ref> input port check box.
The Frame Conversion block does not make any changes to the input signal other
than the sampling mode. In particular, the block does not re buffer or resize 2-D inputs.
Because 1-D vectors cannot be frame based, when the input is a length-M 1-D vector and
the block is in Frame-based mode, the output is a frame-based M-by-1 matrix — that is, a
single channel.
3.5 INTEGER TO BIT CONVERTER
The Integer to Bit Converter block maps each integer or fixed-point value in the
input vector to a group of bits in the output vector. The block maps each integer value (or
stored integer when a fixed point input is used) to a group of M bits, using the selection
for the Output bit order to determine the most significant bit. The resulting output vector
length is M times the input vector length.
For integer-valued inputs, if M is the Number of bits per integer and Treat input
values as is set to Unsigned, then the input values must be between 0 and 2M-1. If M is the

Number of bits per integer and Treat input values as is set to Signed, then the block maps
each group of M bits to an integer between –2M-1 and 2M-1-1.
This block is single-rate and single-channel. The input can be either a scalar or a
frame-based column vector. For a sample-based scalar input, the output is a 1-D signal
with M elements. For a frame-based column vector input, the output is a frame-based
column vector with a length equal to M times the input signal length.
A simulation-only runtime input value check is performed. If any input value is

outside of the Number of bit per integer value range, the block will issue an error. The
block can accept the data types int8, uint8, int16, uint16, int32, uint32, single, double,
and fixed point.
3.6 BIT TO INTEGER CONVERTER
The Bit to Integer Converter block maps groups of bits in the input vector to
integers in the output vector.
For unsigned integers, if M is the Number of bits per integer, then the block maps
each group of M bits to an integer between 0 and 2 M-1. As a result, the output vector
length is 1/M times the input vector length. For signed integers, if M is the Number of
bits per integer , then the block maps each group of M bits to an integer between –2M-1
and 2M-1-1.
If the input signal is a sample-based input, then it must be a vector whose length equals
the value for the Number of bits per integer parameter. If the input is frame-based, then it
must be a column vector whose length is an integer multiple of Number of bits per
integer.
3.7 VECTOR SCOPE

The Vector Scope block is a comprehensive display tool similar to a digital

oscilloscope. The block can display time-domain, frequency-domain, or user-defined
signals. You can use the Vector Scope block to plot consecutive time samples from a
frame-based vector, or to plot vectors containing data such as filter coefficients or
spectral magnitudes. To compute and plot the period gram of a signal with a single block,
use the Spectrum Scope block.
The input to the Vector Scope block can be any real-valued M-by-N matrix,
column or row vector, or 1-D (un oriented) vector, where 1-D vectors are treated as
column vectors. Regardless of the input frame status, the block treats each column of an
M-by-N input as an independent channel of data with M consecutive samples. The block
plots each sample of each input channel sequentially across the horizontal axis of the
plot.
3.7.1 SCOPE PROPERTIES
The Scope Properties pane enables you to plot time-domain, frequency-domain,

or user-defined data, and adjust the horizontal display span of the plot. The scope
displays frames of data, and updates the display for each new input frame.
The Input domain parameter specifies the domain of the input data. If you select
Time, for M-by-N inputs containing time-domain data, the block treats each of the N
input frames (columns) as a succession of M consecutive samples taken from a time
series. That is, each data point in the input frame is assumed to correspond to a unique
time value. Also, the Time display span (number of frames) parameter appears on the
pane. Enter a scalar value greater than or equal to one that corresponds to the number of
frames to be displayed across the width of the scope window.
If you select Frequency for the Input domain parameter, for M-by-N inputs
containing frequency-domain data, the block treats each of the N input frames (columns)
as a vector of spectral magnitude data corresponding to M consecutive ascending
frequency indices. That is, when the input is a single column vector, u, each value in the

input frame, u(i), is assumed to correspond to a unique frequency value, f(i), where
f(i+1)>f(i).
If you select User-defined for the Input domain parameter, the block does not
assume that the input frame data is time-domain or frequency-domain data. You can plot
the data in the appropriate manner. Also, the Horizontal display span (number of frames)
parameter appears on the plane. Enter a scalar value greater than or equal to one that
corresponds to the number of frames to be displayed across the width of the scope
window.
3.8 PORT SELECT
The Input Port block is a connecting port from the Simulink, or mathematical,
part of the model to an RF physical part of the model. The Input Port block lets you
provide the parameter data needed to calculate the modeling frequencies and the
baseband-equivalent impulse response for the physical subsystem. It also lets you specify
information about how to interpret the incoming Simulink signal. For more information
about how the Input Port block converts the mathematical Simulink signals to RF Block
set physical modeling environment signals. When you set the Select parameter to Rows,
the block uses the one-dimensional indices you specify to select matrix rows, and all
elements on the chosen rows are included. When you set the Select parameter to
Columns, the block uses the one-dimensional indices you specify to select matrix
columns, and all elements on the chosen columns are included. A given input row or
column can appear any number of times in any of the outputs, or not at all.
CHAPTER 4

QUADRATURE AMPLITUDE MODULATION (QAM)
4.1 INTRODUCTION
Towards the end of the 1950s there was a considerable amount of interest in
digital phase modulation transmission schemes as an alternative to digital amplitude
modulation. Digital phase modulation schemes are those whereby the amplitude of the
transmitted carrier is held constant but the phase changed in response to the modulating
signal. Such schemes have constellation diagrams of the form shown in Figure. It was a
natural extension of this trend to consider the simultaneous use of both amplitude and
phase modulation. The first paper to suggest this idea was by C.R. Cahn in 1960, who
described a combined phase and amplitude modulation system.
Fig4.1: square QAM constellation
Then its simply extended phase modulation to the multilevel case by allowing
there to be more than one transmitted amplitude at any allowed phase. This had the

effect of duplicating the original phase modulation or phase shift keying (PSK)
constellation which essentially formed a circle. Such duplication led to a number of
concentric circles depending on the number of amplitude levels selected. Each circle had
the same number of phase points on each of its rings. Only Gaussian channels
characteristic of telephone lines impaired by thermal noise were considered. Using a
series of approximations and a wholly theoretical approach, he came to the conclusion
that these amplitude and phase modulation (AM-PM) systems allowed an increased
throughput compared to phase modulation systems when 16 or more states were used and
suggested that such a system was practical to construct.
4.1.1 QAM Implementations
The Japanese started to show interest in QAM schemes as they considered they
might have application in both satellite and microwave radio links. In 1976 Miyauchi,
Seki and Ishio published a paper devoted to implementation techniques . They considered
implementation by superimposing two 4-level PSK modulation techniques at different
amplitudes to achieve a square QAM constellation and using a similar process in reverse
at the demodulator, giving them the advantage of being able to use existing PSK
modulator and demodulator circuits.
This method of implementing QAM .They implemented a prototype system

without clock or carrier recovery and concluded that its performance was sufficiently
good to merit further investigation. Further groundwork was covered in 1978 by
W.Weber, again working on satellite applications, who considered differential (i.e. non-
coherent) encoding systems for the various constellations still in favour. His paper
essentially added a theoretical basis to the differential techniques that had been in use at
that point, although suggesting that non-coherent demodulation techniques deserved more
attention.

Fig 4.2: Optimum 32-level constellation
The construction of QAM prototype systems worldwide started to emerge. A

paper from the CNET laboratories in France by Dupuis et al. [16] considered a 140
Mbit/s modem for use in point-to-point links in the 10-11 GHz band. This prototype
employed the square 16-level constellation and included carrier recovery, although no
details were given. Theoretical calculations of impairments were presented followed by
measurements made over a 58 km hop near Paris. Their conclusions were that QAM had
a number of restrictions in its use, relating to its sensitivity to non-linearities, and that in
the form they had implemented it, PSK offered improved performance. However, they
suggested that with further work these problems might be overcome.
The Japanese simultaneously announced results from a prototype 200 Mbit/s 16-
QAM system in a paper by Horikawa, Murase and Saito. They used differential coding

coupled with a new form of carrier recovery based on a decision feedback method. Their
modem was primarily designed for satellite applications and their experiment included
the use of TWT amplifiers, but was only carried out back-to-back in the laboratory. Their
conclusions were that their prototype had satisfactory performance and was an efficient
way to increase bandwidth efficiency. One of the last of the purely theoretical, as
opposed to practical papers on QAM appeared in April 1980, marking the progression of
QAM from a technical curiosity into a practical system, some twenty years after its
introduction. This came from V. Prabhu of Bell Labs , further developing the theory to
allow calculation of error probabilities in the presence of co-channel interference. Prabhu
concluded that 16-QAM had a co-channel interference immunity superior to 16-PSK but
inferior to 8-PSK.
4.1.2 QAM for channel Communications
Considering QAM for voice transmission over Rayleigh fading channels, the first
major paper considering QAM for mobile radio applications. In these papers, it was
recognized that when a Gray code mapping scheme was used, some of the bits
constituting a symbol had different error rates from other bits. Gray coding is a method of
assigning bits to be transmitted to constellation points in an optimum manner . For the
16-level constellation two classes of bits occurred, for the 64-level three classes and so
on.Efficient mapping schemes for pulse code modulated (PCM) speech coding were
discussed where the most significant bits (MSBs) were mapped onto the class with the
highest integrity. A number of other schemes including variable threshold systems and
weighted systems were also discussed. Simulation and theoretical results were compared
and found to be in reasonable agreement. They used no carrier recovery, clock recovery
or AGC, assuming these to be ideal, and came to the conclusion that channel coding and
post-enhancement techniques would be required to achieve acceptable performance.
This work was continued, resulting in a publication in 1990 by Hanzo, Steele and
Fortun , again considering QAM for mobile radio transmission, where again a theoretical
argument was used to show that with a Gray encoded square constellation, the bits

encoded onto a single symbol could be split into a number of subclasses, each subclass
having a different average BER. The authors then showed that the difference in BER of
these different subclasses could be reduced by constellation distortion at the cost of
slightly increased total BER, but was best dealt with by using different error correction
powers on the different 16- QAM subclasses. A 16 kbit/s sub-band speech coder was
subjected to bit sensitivity analysis and the most sensitive bits identified were mapped
onto the higher integrity 16-QAM subclasses, relegating the less sensitive speech bits to
the lower integrity classes.
Furthermore, different error correction coding powers were considered for each
class of bits to optimize performance. Again ideal clock and carrier recovery were used,
although this time the problem of automatic gain control (AGC) was addressed. It was
suggested that as bandwidth became increasingly congested in mobile radio, microcells
would be introduced supplying the required high SNRs with the lack of bandwidth being
an incentive to use QAM.
Further QAM schemes for hostile fading channels characteristic of mobile

telephony can be found in the following recent references. If Fehers previously
mentioned NLA concept cannot be applied, then power-inefficient class A or AB linear
amplification has to be used, which might become an impediment in lightweight, low-
consumption handsets. However, the power consumption of the low-efficiency class A
amplifier is less critical than that of the digital speech and channel codes. In many
applications 16- QAM, transmitting 4 bits per symbol reduces the signalling rate by a
factor of 4 and hence mitigates channel dispersion, thereby removing the need for an
equalizer. while the higher SNR demand can be compensated by diversity reception.
Significant contributions were made by Cavers, Stapleton et al. at Simon Fraser
University, Burnaby, Canada in the field of pre- and post-distorter design.
Out-of-band emissions due to class AB amplifier non-linearities and hence

adjacent channel interferences can be reduced by some 15-20 dB using Stapleton’s
adaptive predistorter and a class AB amplifier with 6 dB back-off, by adjusting the

predistorter’s adaptive coefficients using the complex convolution of the predistorter’s

input signal and the amplifier’s output signal. Further aspects of linearised power
amplifier design are considered in references.
4.2 History of OFDM-based QAM
The first QAM-related so-called orthogonal frequency division multiplexing

(OFDM) scheme was proposed by Chang in 1966 [142] for dispersive fading channels,
which has also undergone a dramatic evolution due to the efforts of Weinstein, Peled,
Ruiz, Hirosaki, Kolb, Cimini, Schüssler, Preuss, Rückriem, Kalet et al.OFDM was
standardised as the European digital audio broadcast (DAB) as well as digital video
broadcast (DVB) scheme.
It constituted also a credible proposal for the recent third-generation mobile radio
standard competition in Europe. It was recently selected as the high performance local
area network (HIPERLAN) transmission technique. The system’s operational principle is
that the original bandwidth is divided in a high number of narrow sub-bands, in which the
mobile channel can be considered non-dispersive. Hence no channel equaliser is required
and instead of implementing a bank of sub-channel modems they can be conveniently
implemented by the help of a single fast fourier Transformer(FFT).
These OFDM systems - often also termed as frequency division multiplexing

(FDM) or multi-tone systems - have been employed in military applications since the
1960s, for example by Bello, Zimmerman, Powers and Zimmerman, Chang and Gibby
and others. Saltzberg studied a multi-carrier system employing orthogonal time–
staggered quadrature amplitude modulation (O-QAM) of the carriers.
The employment of the discrete Fourier transform (DFT) to replace the banks of
sinusoidal generators and the demodulators was suggested by Weinstein and Ebert in
1971, which significantly reduces the implementation complexity of OFDM modems. In
1980, Hirosaki suggested an equalisation algorithm in order to suppress both inter symbol

and inter subcarrier interference caused by the channel impulse response or timing and
frequency errors. Simplified OFDM modem implementations were studied by Peled in
1980, while Hirosaki introduced the DFT based implementation of Saltzberg’s O-QAM.
OFDM system. From Erlangen University, Kolb, Schuler, Peruses and Rückriem
[conducted further research into the application of OFDM. Cimini and Kalet published
analytical and early seminal experimental results on the performance of OFDM modems
in mobile communications channels. More recent advances in OFDM transmission were
presented in the impressive state-of the- art collection of works edited by Fazel and
Fettweis, including the research by Fettweis at Dresden University, Rohling et al. at
Braunschweig University, Vandendorp at Loeven University, Huber et al. at Erlangen
University, Lindner et al. at Ulm University, Kammeyer et al. at Brehmen University and
Meyr et al. at Aachen University, but the individual contributions are too numerous to
mention.While OFDM transmission over mobile communications channels can alleviate
the problem of multipath propagation, recent research efforts have focused on solving a
set of inherent difficulties regarding OFDM, namely the peak–to–mean power ratio, time
and frequency synchronisation, and on mitigating the effects of the frequency selective
fading channel.
4.2.1 Ofdm
Combining OFDM transmissions with code division multiple access (CDMA)

allows us to exploit the wideband channel’s inherent frequency diversity by spreading
each symbol across multiple subcarriers. This technique has been pioneered by Yee,
Linnartz and Fettweis , by Chouly, Brajal and Jourdan , as well as by Fettweis, Bahai and
Anvari . Fazel and Papke investigated convolutional coding in conjunction with
OFDM/CDMA. Prasad and Hara compared various methods of combining the two
techniques, identifying three different structures, namely multi-carrier CDMA (MC-
CDMA), multi-carrier direct sequence CDMA (MC-DS-CDMA) and multi-tone CDMA
(MT-CDMA). Like non-spread OFDM transmission, OFDM/CDMA methods suffer

from high peak-tomean power ratios, which are dependent on the frequency domain
spreading scheme, as investigated by Choi, Kuan and Hanzo.
4.2.2 History of QAM-based coded modulation
The history of channel coding or Forward Error Correction (FEC) coding dates
back to Shannon’s pioneering work in 1948, in which he showed that it is possible to
design a communication system with any desired small probability of error, whenever the
rate of transmission is smaller than the capacity of the channel.
While Shannon outlined the theory that explained the fundamental limits imposed
on the efficiency of communications systems, he provided no insights into how to
actually approach these limits. This motivated the search for codes that would produce
arbitrarily small probability of error. Specifically, Hamming and Golay were the first to
develop practical error control schemes. Convolutional codes were later introduced by
Elias in 1955, while Viterbi invented a maximum likelihood sequence estimation
algorithm in 1967 for efficiently decoding convolutional codes. In 1974, Bahl proposed
the more complex Maximum A-Posteriori (MAP) algorithm,
Which is capable of achieving the minimum achievable BER. The first successful
application of channel coding was the employment of convolutional codes in deep-space
probes in the 1970s. However, for years to come, error control coding was considered to
have limited applicability, apart from deep-space communications. Specifically, this is a
power-limited scenario, which has no strict bandwidth limitation.
By contrast mobile communications systems constitute a power- and bandwidth-

limited scenario. In 1987, a bandwidth efficient Trellis Coded Modulation (TCM)
scheme employing symbol-based channel interleaving in conjunction with Set-
Partitioning (SP) assisted signal labelling was proposed by Ungerböck. Specifically, the

TCM scheme, which is based on combining convolutional codes with multidimensional

signal sets, constitutes a bandwidth efficient scheme that has been widely recognised as
an efficient error control technique suitable for applications in mobile communications.
Another powerful coded modulation scheme utilising bit-based channel

interleaving in conjunction with Gray signal labelling, which is referred to as Bit-
Interleaved Coded Modulation (BICM), was proposed by Zehavi as well as by Caire,
Taricco and Biglieri. Another breakthrough in the history of error control coding is the
invention of turbo codes by Berrou, Glavieux and Thitimajshima in 1993. Convolutional
codes were used as the component codes and decoders based on the MAP algorithm were
employed. The results proved that a performance close to the Shannon limit can be
achieved in practice with the aid of binary codes. The attractive properties of turbo codes
have attracted intensive research in this area. As a result, turbo coding has reached a state
of maturity within just a few years and was standardised in the recently ratified third-
generation (3G) mobile radio systems.
4.2.3 Adaptive qam optimisation for ofdm
We have considered the design trade-offs of adaptive versus space-time coded

transmissions. Although both methods aim for mitigating the effects of fading-induced
channel quality fluctuations, their approach is fundamentally different. Adaptive QAM
aims for adjusting the modulation modes for the sake of maintaining the target integrity,
despite the channel quality fluctuations. By contrast, space-time coding employs multiple
transmitters and receivers for the sake of directly mitigating the channel quality
undulations.
It transpired that both techniques were capable of attaining a similar performance,

although AQAM imposed a lower complexity owing to employing a single transmitter
and receiver. In this chapter our discourse evolves further by providing a comparative
study of the various approaches developed for optimising the AQAM switching
thresholds and then employing the thresholds in the context of both AQAM and space-

time coded OFDM as well as frequency-domain spread MC-CDMA. Let commence our
detailed discourse with a glimpse of history. In recent years the concept of intelligent
multi-mode, multimedia transceivers (IMMT) has emerged in the context of wireless
systems.
The range of various existing solutions that have found favor in already
operational standard systems was summarized in the excellent overview by Nanda et al. .
The aim of these adaptive transceivers is to provide mobile users with the best possible
compromise amongst a number of contradicting design factors, such as the power
consumption of the hand-held portable station (PS), robustness against transmission
errors, spectral efficiency, tele traffic capacity, audio/video quality and so forth .
4.3 QAM and QPSK
Fig4.3: Quadrature modulator
The motivation for QAM comes from the fact that a DSBSC signal occupies
twice the bandwidth of the message from which it is derived. This is considered wasteful

of resources. QAM restores the balance by placing two independent DSBSC, derived
from message #1 and message #2, in the same spectrum space as one DSBSC. The
bandwidth imbalance is removed.
In digital communications this arrangement is popular. It is used because of its

bandwidth conserving (and other) properties. It is not used for multiplexing two
independent messages. Given an input binary sequence (message) at the rate of n bit/s,
two sequences may be obtained by splitting the bit stream into two paths, each of n/2
bit/s. This is akin to a serial-to-parallel conversion.
Fig4.4: QPSK modulator
4.3.1 QAM becomes QPSK

The QAM modulator is so named because, in analog applications, the messages

do in fact vary the amplitude of each of the DSBSC signals. In QPSK the same modulator
is used, but with binary messages in both the I and Q channels, as described above. Each
message has only two levels, ±V volt. For a non-band limited message this does not vary
the amplitude of the output DSBSC. As the message changes polarity this is interpreted
as a 1800 phase shift, given to the DSBSC.
Thus the signal in each arm is said to be undergoing a 1800 phase shift, or phase
shift keying - or PSK. Because there are two PSK signals combined, in quadrature, the
two-channel modulator gives rise to a quadrature phase shift keyed - QPSK - signal.
Viewed as a phasor diagram (and for a non-band limited message to each channel), the
signal is seen to occupy any one of four point locations on the complex plane. These are
at the corner of a square (a square lattice), at angles π/4, 3π/4, 5π/4 and 7π/4 to the real
axis.
4.3.2 M-psk and M-qam
The above has described digital-QAM or QPSK. This signal is also called 4-PSK
or 4- QAM. More generally signals can be generated which are described as M-QAM or
M-PSK. Here M = 2L, where L = the number of levels in each of the I and Q arms. For
the present experiment L = 2, and so M = 4. The ‘M’ defines the number of points in the
signal constellation. For the cases M > 4 then M-PSK is not the same as M-QAM.
4.4 THE QAM RECEIVER
The QAM receiver follows the similar principles to those at the transmitter, and is
illustrated in idealised from in the block diagram of Figure 3. It is idealised because it
assumes the incoming signal has its two DSBSC precisely in phase quadrature. Thus only
one phase adjustment is required.

Fig4.5: QAM demodulator for QPSK
The parallel-to-serial converter block performs the following operations:

 Regenerates the bit clock from the incoming data.
 Regenerates a digital waveform from both the analog outputs of the I and
Q arms.
 Re-combines the I and Q signals, and outputs a serial data stream.
Not shown is the method of carrier acquisition. This ensures that the oscillator,
which supplies the local carrier signal, is synchronized to the received (input) signal in
both frequency and phase. In this experiment we will use a stole carrier to ensure that
carrier signal in the transmitter and receiver are in synchronism with each other. (Please
read about Costas Receiver to understand more about carrier acquisition). In this
experiment, two independent data sequences will be used at the input to the modulator,
rather than having digital circuitry to split one data stream into two (the serial-to-parallel

converter). Two such independent data sequences, sharing a common bit clock (2.083
kHz), are available from a single sequence generator module. The data stream from
which these two channels are considered to have been derived would have been at a rate
of twice this - 4.167 kHz.
Low pass filter band limiting and pulse shaping is not a subject of enquiry in this
experiment. So a single band pass filter at the adder (summer) output will suffice,
providing it is of adequate bandwidth. A 100 kHz channel filters module is acceptable.
4.5 QAM TRANSMITTER IMPLEMENTATION
QAM is widely used method for transmitting digital data over bandpass channels. A
basic QAM transmitter
Fig 4.6: QAM Transmitter
Complex Symbols

 Quadriphase-Shift Keying (QPSK)

 4-QAM
 Transmitted signal is contained in the phase
Fig 4.7: pilot symbols
4.6 QAM MODULATION
The simplest type of digital modulation involves transmitting a sequence of

waveforms (“symbols”) s(t) of equal duration T where each waveform is chosen
independently from a set of M. This allows us to transmit up to b = log2 (M) bits per
symbol. A common sets of such symbols are those where the real and imaginary parts of
the complex baseband signal are each modulated in amplitude. This known as Quadrature
Amplitude Modulation (QAM).

Each of the possible symbols is a combination of a real and a complex value and
can be plotted as a point in the complex plane. Each point represents one of the M
possible symbols. This plot is called a constellation. For example, the plots below show
the constellations used by the 802.11n WLANs as given in the IEEE standard:
QAM modulation with M symbols is known as M-QAM, for example 16-QAM

etc. The constellations for M = 2 and M = 4 are the same as those for BPSK and QPSK
modulation as explained below. Values of M from 2 up to 1024 (10 bits per symbol) or
even more are used. Higher values of M are used on channels with low levels of noise
and distortion. Constellation sizes that are even powers of 2 (M = 2;4;16;64; : : :)are
typically used to make the constellation the same in both axes and simplify
implementation. However, non-square constellations are also used for low values of M or
where maximum power efficiency is desired. For example, here is an example of a non-
square constellation for M = 8 (b = bits/symbol). PAM signals occupy twice the
bandwidth required for the baseband Transmit two PAM signals using carriers of the
same frequency but in phase and quadrature.
4.7 QAM DEMODULATION
 Exact knowledge of the carrier and symbol clock phases and frequencies.
 Apply the Hilbert Transform of the received signal to generate the pre-
envelope s+(t)
∞
S(t) = s+¿¿ (t) e− j w t =
c
∑ ¿ ¿+ j b k) gt (t-k T) (4.7a)
k=−∞
~s (n T ) = a n + j b n (4.7b)
QAM demodulator using the complex envelope

Fig 4.8: QAM demodulator
QAM demodulation by modulation then filtering
 Construct in-phase i(t) and quadrature q(t) signals

 Low pass filter them to obtain baseband signals a(t) and b(t)
4.8 AWGN Channel
The AWGN channel is the simplest channel model used in most communication
systems. The thermal noise in the receivers can be characterized as an additive white
Gaussian process. Although there are other factors inducing channel noise, such as
antenna temperature, receiver filter, and multipath fading, only multipath fading will be
studied in this chapter.

The AWGN Channel block adds white Gaussian noise to a real or complex input
signal. When the input signal is real, this block adds real Gaussian noise and produces a
real output signal. When the input signal is complex, this block adds complex Gaussian
noise and produces a complex output signal. This block inherits its sample time from the
input signal.
This block uses the Signal Processing Block set Random Source block to generate
the noise. Random numbers are generated using the Ziggurat method, which is the same
method used by the MATLAB random function. The Initial seed parameter in this block
initializes the noise generator. Initial seed can be either a scalar or a vector whose length
matches the number of channels in the input signal. For details on Initial seed. The signal
inputs can only be of type single or double. The port data types are inherited from the
signals that drive the block.
AWGN channel adds white Gaussian noise to the signal that passes through it.
The AWGN block can process either sample-based or frame-based data, and it lets you
specify the variance of the noise in one of four ways:
 Directly as a mask parameter
 Directly as an input signal
 Indirectly via a signal-to-noise ratio parameter
 Indirectly via an Es/No parameter
4.8.1 Awgn
Add white gaussian noise to signal

Syntax
y = awgn(x,snr)
y = awgn(x,snr,sigpower)
y = awgn(x,snr,'measured')
y = awgn(x,snr,sigpower,state)
y = awgn(x,snr,'measured',state)
y = awgn(power type)
Description:
 y = awgn(x,snr) adds white gaussian noise to the vector signal x. the scalar snr
specifies the signal-to-noise ratio per sample, in db. if x is complex, awgn adds
complex noise. this syntax assumes that the power of x is 0 dbw.
 y = awgn(x,snr,sigpower) is the same as the syntax above, except that sigpower is

the power of x in dbw.
 y = awgn(x,snr,'measured') is the same as y = awgn(x,snr), except that awgn

measures the power of x before adding noise.
 y = awgn(x,snr,sigpower,state) is the same as y = awgn(x,snr,sigpower), except

that awgn first resets the state of the normal random number generator randn the
integer state.
 y = awgn(x,snr,'measured',state) is the same as y = awgn(x,snr,'measured'), except

that awgn first resets the state of normal random number generator randn to the
integer state.
 y = awgn(power type) is the same as the previous syntaxes, except that the string
power type specifies the units of snr and sign power. choices for power type are
'db' and 'linear'. if power type is 'db', then snr is measured in db and sign power is

measured in dbw. if power type is 'linear', snr is measured as a ratio and sign
power is measured in watts.
4.8.2 White gaussian noise
The WGN function generates random matrices using a white gaussian noise
distribution. you specify the power of the noise in either dbw (decibels relative to a watt),
dbm, or linear units. you can generate either real or complex noise. for example, the
command below generates a column vector of length 50 containing real white gaussian
noise whose power is 2dbw. the function assumes that the load impedance is 1 ohm.
To generate complex white gaussian noise whose power is 2 watts, across a load
of 60 ohms, use either of the commands below. the ordering of the string inputs does not
matter. to send a signal through an additive white gaussian noise channel, use the awgn
function.
4.9 FADING CHANNEL
Channel fading is generally categorized into large-scale and small-scale fading,

which often occur simultaneously. Large-scale fading results from shadowing terrain
contours such as hills, forests, or buildings, relative to the distance between transmitter
and receiver. Small-scale fading, also known as Rayleigh fading, is not determined by the
distance in communication. The small-scale fading is manifest in two ways: the signal
spreading and the time variation.
4.9.1 COMPENSATING FOR FADING
 A communication system involving a fading channel usually requires

component (s) that compensate for the fading. here are some typical
approaches:
 Differential modulation or a one-tap equalizer can help compensate for a

frequency-flat fading channel.

 An equalizer with multiple taps can help compensate for a frequency-selective

fading channel.
 describing the noise level of an awgn channel
 The relative power of noise in an awgn channel is typically described by
quantities such as
 Signal-to-noise ratio (snr) per sample. this is the actual input parameter to the
awgn function.
 Ratio of bit energy to noise power spectral density (e b/n0). this quantity is used
by bertool and performance evaluation functions in this toolbox.
 Ratio of symbol energy to noise power spectral density (es/n0)
Relationship between es/n0 and eb/n0 The relationship between es/n0 and eb/n0, both
expressed in db, is as follows:
ES/NO(db)=Eb/N0(db)+10log10(K) . (4.9.1a)
where k is the number of information bits per symbol.
In a communication system, k might be influenced by the size of the modulation

alphabet or the code rate of an error-control code. for example, if a system uses a rate-1/2
code and 8-psk modulation, then the number of information bits per symbol (k) is the
product of the code rate and the number of coded bits per modulated symbol: (1/2)
log2(8) = 3/2. in such a system, three information bits correspond to six coded bits, which
in turn correspond to two 8-psk symbols.
Derivation for complex input signals.
You can derive the relationship between es/n0 and snr for complex input signals as
follows:
ES/NO(db) =10log10(S.Tsym)/(N/Bn))

=10 log10((TsymFS)(S/N))
=10log10(Tsym/Tsamp)+SNR(db) (4.9.1b)
Where
 s = input signal power, in watts

 n = noise power, in watts
 bn = noise bandwidth, in hertz.
CHAPTER 5
CHANNEL CODING AND ISI
5.1 CHANNEL CODING
Since C. E. Shannon’s pioneering work on mathematical theory for digital

communications in 1948-49, the subject of error control coding has emerged as a
powerful and practical means of achieving efficient and reliable communication of
information in a cost effective manner. Suitability of numerous error control schemes in
digital transmission systems, wire line and wireless, has been studied and reported in
detail in the literature.
The major categories of activities on error control coding can broadly be identified as the
following:
a) To find codes with good structural properties and good asymptotic error
performance.
b) To devise efficient encoding and decoding strategies for the codes.
c) To explore the applicability of good coding schemes in various digital

transmission and storage systems and to evaluate their performance.

The encoding operation for a (n, k) error control code is a kind of mapping of
sequences, chosen from a k-dimensional subspace to a larger, n-dimensional vector space
of n-tuples defined over a finite field and with n > k. Decoding refers to a reverse
mapping operation for estimating the probable information sequence from the knowledge
of the received coded sequence. If the elements (bit, or a symbol made of group of bits)
of the message sequence at the input to the encoder are defined over a finite field of qi
elements and the sequence elements at the output of the encoder are defined over (same
or a different) finite field with q o elements.
For a binary code, q i = q 0 = 2 and hence, R = Lin / Lout. A (7,4) Hamming code
is an example of a binary block code whose rate R = 4/7. This code will be addressed
later in greater detail. For an error correction code, R < 1.0 and this implies that some
additional information (in the form of ‘parity symbol’ or ‘redundant symbol’) is added
during the process of encoding. This redundant information, infused in a controlled way,
help us in decoding a received sequence to extract a reliable estimate of the information
bearing sequence. Now, it is interesting to note that the purpose of error control can be
achieved in some situations even without accomplishing the complete process of
decoding. Accordingly, the process of error control can be categorized into the following:
5.1.1 Nature of communication channel
Effects of many physical communication channel manifest in random and

isolated errors while some channels cause burst errors. The modulation technique
employed for transmission of information, sensitivity level of a receiver rate of
information transmission is some other issues.
5.1.2 Available channel bandwidth
As mentioned, use of an error-control scheme involves addition of controlled

redundancy to original message. This redundancy in transmitted massage calls for larger
bandwidth than what would be required for an encoded system. This undesirable fact is

tolerable because of the obtainable gains or advantages of coded communication system

over an encoded one for a specified overall system performance in terms of BER or cost.
5.1.3 Hardware complexity cost and delay
Some FEC codes of larger block length asymptotically satisfy the requirements of
high rate as well as good error correcting capability but the hardware complexity,
volume, cost and decoding delay of such decoders may be enormous. For a system
designer, the choice of block length is somewhat limited.
5.1.4 The coding gain
FEC codes of different code rates and block sizes offer different coding gains in
Eb/No over an un coded system. At the first level, the coding gain is defined as:
[( Eb /No in dB needed by a un coded system to achieve a specified BER of 10−x ) –
( Eb /No in dB needed by an FEC coded system to achieve the same BER of 10−x ).]
5.2 BLOCK CODES
The block coder input is a stream of information bits. The coder segments this bit
stream into blocks of k information bits and for each block it calculates a number of r
check bits, or it picks the r check bits from a tabulated set of values. It then transmits the
entire block, or codeword of n = k + r channel bits. This is called an block code.
If errors occur in sufficiently few of these transmitted channel bits, the r check
bits may provide the receiver with sufficient information to enable it to detect and/or
correct the channel errors. The code efficiency (or code rate) is k/n. If the k information
bits are transmitted unaltered first followed by the transmission of the r check bits it is
called a systematic code. A non-systematic block code is one which has the check bits
interspersed between the information bits.

5.2.1 Repetition Codes
These are the simplest type of block codes. One way to detect an error in an
information block is to send the information twice. The two received blocks are
compared bit by bit and if there is a difference an error has occurred. This method may be
extended by sending the information block three times. If one block differs from the other
two, assume an error has occurred in that block and discard it.
This is repetition coding. It is simple to implement but very inefficient in terms of

information transfer. Another method is to encode a single symbol into a block of n
identical bits an (n,1) block code. There are only two code words - the sequence of n 0s
and the sequence of n 1s. The first bit is the information bit, the other r = n − 1 bits are
check bits. The value of each check bit is identical to the information bit.
The decoder might use the following rules:
 Count the number of 0s and the number of 1s in the received bits. If there are
more received 0s than 1s, decide that the all-0 codeword was sent.
 If there are more 1s than 0s decide that the all-1 codeword was sent.
 If the number of 1s equals the number of 0s do not decide - just flag a
decoding failure and perhaps generate and ARQ (automatic request to repeat
the message).
 In this case the Hamming distance is n so that the original data can be
recovered if there are less than (n−1)/2 errors in the received code word. this
is the basis of the 2nd and 3rd steps above. This rule will decode correctly in
all cases where the channel noise changes less than half the bits in any one
block.
If the channel noise changes more than half of the bits in any one block, the
decoder will make a decoding error, i.e., it will decode the received word into the wrong
codeword. If channel errors occur infrequently the probability of a decoding failure or a
decoding error for a repetition code of long block length is very small indeed.

5.2.2 Parity check codes
This is an example of a high information rate (R = k/n) code. It appends a parity

check bit to the end of the information bits. This check bit is the modulo-2 sum (modulo-
2 addition is equivalent to the exclusive OR logical operation) of the codeword (n − 1)
information bits. If the number of 1s in the information word is even, then the modulo-2
sum of all the information bits will be equal to 0. If the number of 1s in the information
word is odd their modulo-2 sum will be equal to 1.
The parity bit is calculated and appended to the information bits to form the
codeword. Even parity means that the parity bit is set so that the total number of 1s in the
codeword is even. Odd parity means that the total number of 1s in the codeword must be
odd. This type of code can only detect and cannot correct errors. A single bit error (or any
number of odd bit errors) will be detected but any combination of two bit errors (or any
number of even bit errors) will cause a decoding error.
Repetitive codes and single parity check codes are, respectively, examples of
extreme and relatively trivial block codes. However, single parity checks are used quite
often with ASCII codes in computer communication
5.3 INTER SYMBOL INTERFERENNCE
Among the different types of distortions in communication Inter Symbol

Interference is a phenomenon that causes the heavy data loss in the communication.
Usually the digital information that is transmitted will be in the form of square waveform
representing the 1’s and 0’s. When this square waveform mixes with the noises and non
linearities in the channel, the square waveform starts to spread and merge with the
adjacent symbol sequence, making the data there to be unreadable.

At the receiver end this data is wrongly decoded as the receiver cannot predict
the correct level of the square waveform leading to the loss of information. ISI is usually
caused due to multipath propagation of the signal in band limited channel and the non-
linear frequency response of the channel. The Fig represents the graphical representation
of ISI.The presence of ISI in the system introduces errors in the decision device at the
receiver output. Therefore, in the design of the transmitting and receiving filters, the
objective is to minimize the effects of ISI,and thereby deliver the digital data to its
destination with the smallest error rate possible.
Fig 5.1: Inter symbol interference
5.3.1 CAUSES OF INTER SYMBOL INTERFERENCE

Multipath propagation
One of the causes of Inter Symbol Interference is multipath propagation in which

a wireless signal from a transmitter reaches the receiver via many different paths. The
causes of this include reflection refraction (such as through the foliage of a tree) and
atmospheric effects such as atmospheric ducting and ionospheric reflection. Since all of
these paths are different lengths - plussome of these effects will also slow the signal
down - this results in the different versions of the signal arriving at different times. This
delay means that part or all of a given symbol will be spread into the subsequent symbols,
thereby interfering with the correct detection of those symbols. Additionally, the various
paths often distort the amplitude and/or phase of the signal thereby causing further
interference with the received signal.
Band-limited channels
Another cause of Inter Symbol Interference is the transmission of a signal through

a band-limited channel, i.e., one where the frequency response is zero above the cutoff
frequency. Passing a signal through such a channel results in the removal of frequency
components above this cutoff frequency; in addition, the amplitude of the frequency
components below the cutoff frequency may also be attenuated by the channel.
Noises and Fading Channels
Noise may be defined as any unwanted signal that interferes with the
communication, measurement or processing of an information-bearing signal. Noise is
present in various degrees in almost all environments. For example, in a digital cellular
mobile telephone system, there may be several variety of noise that could degrade the
quality of communication, such as acoustic background noise, thermal noise,
electromagnetic radio-frequency noise, co-channel interference, radio channel distortion,

echo and processing noise. Noise can cause transmission errors and may even disrupt a
communication process.
A Communication Channel is known as fading channel if in which it has to face

different fading phenomenon, during signal transmission. In real world environment, the
radio propagation effects combine together and multipath is generated by these fading
channels. Due to multiple signal propagation paths, multiple signals will be received by
receiver and the actual received signal level is the vector sum of the all signals . Among
the noises, the following three are the noises that widely affect the digital
communication. They are Additive White Gaussian Noise (AWGN), Rayleigh Fading,
and Rician Fading. These noises and fading cause huge loss of data in digital
communication. And fading causes a major error called Inter Symbol Interference (ISI).
Additive White Gaussian Noise Channel
The simplest type of channel is the Gaussian channel. It is often referred to the
additive white Gaussian noise (AWGN) channel. Basically, it is the noise generated in
the receiver side if we assume that the transmitter is ideal and noiseless. This type of
noise is assumed to have a constant power spectral density over the whole channel
bandwidth and its amplitude Probability Density Function (PDF) obeys the statistics of a
Gaussian distribution. Gaussian noise is very important in the analysis of communication
system performance. The classical AWGN channel is always considered as the starting
point to develop basic systems performance
Rayleigh Fading Phenomenon
In a mobile communication system, one of the most devastating phenomena is

fading. Fading is the direct result of multi-path propagation where signals propagate
along different paths before arriving at the receiver antenna. Rayleigh fading occurs when
no Line-of-Sight (LOS) path exists in between transmitter and receiver, but only have
indirect path, then the resultant signal received at the receiver will be the sum of all the

reflected and scattered waves . These radio signals may arrive at receiver after different
delays, with different amplitudes, and with different phases. Because there are so many
different received signal components, constructive and destructive interference results in
fading. Thus, in a multi-path fading channel, when a signal pulse is transmitted, the
receiver receives the superposition of many pulses. These multi-paths cause a wide
fluctuation in the received signal magnitude, which makes reliable transmission of
information a challenge.
Rician Fading Phenomenon
The Rician fading phenomenon is very much similar to the Rayleigh fading
phenomenon. The difference is that it occurs when there is a LOS as well as the non-LOS
path in between the transmitter and receiver, i.e. the received signal comprises on both
the direct and scattered multipath waves. This phenomenon also causes huge losses due
to signal collision and interference making communication degraded.
5.3.2 Effect of Channel Bandwidth Limitation on ISI
Consider a baseband digital signal with symbol rate Rs symbols/second that is

composed of a sum of square pulses and that is transmitted through different baseband
channels with different bandwidths:
Channel with Infinite Bandwidth
Such a channel passes all signal components. In this case, the received signal will
be exactly the same as the transmitted square wave since the complete signal is passed.
So, the transmitted data will not experience any ISI at all.

Fig 5.2 Channel with Infinite Bandwidth
Wideband Channel with channel larger than Rs/2
The bandwidth of the channel in this case is wide but not infinite, so a relatively
large amount of the signal power will pass and a small amount at high frequencies will be
rejected. The data in this case experiences some ISI but data can easily be recovered since
the ISI is limited.

Fig 5.3: Wideband Channel with channel larger than Rs/2
Channel Bandwidth is Equal to One Half Symbol Rate Rs/2
The first null (zero) in the power spectrum density of transmitted data occurs at
one half the sample rate Rs/2. The received signal in this situation experiences significant
amount of ISI. However, the data is still recoverable using some signal processing
algorithms. This represents the minimum channel bandwidth that would allow us to
recover the data completely. Any channel bandwidth below this would cause a problem.

Fig 5.4: Channel Bandwidth is Equal to One Half Symbol Rate Rs/2
Channel Bandwidth is Lower than One Half Symbol Rate Rs/2
In this case, the ISI is huge and loss of data will occur. It is not possible to recover
back the data completely no matter what signal processing algorithms are used. We see
from the previous 4 cases that when transmitting square pulses and the bandwidth of the
channel is not infinite, then ISI will occur. However, as long as the bandwidth of the
channel is greater than one half the symbol rate, data can be recovered but possibly using
some signal processing algorithms to remove the effect of ISI. If the channel bandwidth
is less than that, then loss of data will certainly occur.

Fig 5.5: Channel Bandwidth is Lower than One Half Symbol Rate Rs/2
5.4 TYPES OF PULSE SHAPING FILTERS
There are mainly four types of filters that are commonly used in communication systems.
Boxcar filters
A filter is said to be a boxcar filter if it employs boxcar function which is zero

over the entire real line except for a single interval where it is equal to a constant, A; it is
a simple step function The boxcar function can be expressed in terms of the uniform
distribution as Boxcar(x) = (b-a) A f (a, b; x) = H(x-a) – H (x-b), where f (a, b; x) is the
uniform distribution of x for the interval [a, b] and H(x) is the Heaviside step function. In
the boxcar function is represented. As with most such discontinuous functions, there is a
question of the value at the transition points. These values are probably best chosen for
each individual application. When a boxcar function is selected as the impulse response

of a filter, the result is a moving average filter. The function is named after its
resemblance to a boxcar, a type of railroad car. The boxcar filter results in infinitely wide
bandwidth for the signal. Thus its usefulness is limited, but it is used widely in wired
baseband communications, where the channel has some extra bandwidth and the
distortion created by the channel can be tolerated.
Fig 5.6: Boxcar function
Sync Filters
In signal processing, a sync filter is an idealized filter that removes all frequency
components above a given cutoff frequency, without affecting lower frequencies, and has
linear phase response. The filter's impulse response is a sinc function in the time domain,
and its frequency response is a rectangular function. It is an ideal low-pass filter in the
frequency sense, perfectly passing low frequencies, perfectly cutting high frequencies;
and thus may it be regarded to be a brick-wall filter. Theoretically the best pulse shaping
filter would be the sync filter, but it cannot be implemented precisely. In Fig, The Sync
function is represented. It is a non-causal filter with relatively slowly decaying tails. It is
also problematic from a synchronization point of view as any phase error results in
steeply increasing Inter Symbol Interference.
Fig 5.7: The sync function
Gaussian Filter
Gaussian filters are significantly better than RRC filters. Gaussian waveform,
which is optimum in terms of energy concentrations in time and frequency domains, has
a large ISI. Pulse shaping filters are normally implemented as oversampled finite impulse
response (FIR) digital filters. In signal processing, oversampling is the process of
sampling a signal with a sampling frequency significantly higher than twice the
bandwidth or highest frequency of the signal being sampled. Oversampling helps avoid
aliasing, improves resolution and reduces noise. The FIR Gaussian pulse shaping filter
design is done by truncating a sampled version of the continuous-time impulse response
of the Gaussian.
Raised cosine filter
Raised-cosine filters are practical to implement and they are in wide use. They
have a configurable excess bandwidth, so communication systems can choose a tradeoff
between a simpler filter and spectral efficiency. The raised-cosine filter is a filter
frequently used for pulse-shaping in digital modulation due to its ability to minimize Inter

Symbol Interference (ISI). Its name is derived from the fact that the non-zero portion of
the frequency spectrum of its simplest form (β=1) is a cosine function, 'raised' up to sit
above the f (horizontal) axis. The efficiency of the filter varies with the value of Roll-off
factor (β) which varies from ‘0’ to ‘1’. In Fig 5, impulse response representation of the
filter for different values of β is shown.
Fig 5.8: Amplitude and frequency response of raised cosine filter
Fig 5.9: Impulse response of raised cosine filte

CHAPTER 6
ADVANTAGES AND APPLICATIONS OF OFDM
The primary advantage of OFDM over single-carrier modulation schemes is its

ability to cope with severe channel conditions without any complex equalization filters. It
is robust against attenuation of high frequencies in a long copper wire, inter carrier
interference and frequency selective fading due to multipath propagation. Channel
equalization in OFDM system is simplified because OFDM uses many slowly modulated
narrow band signals rather than using one rapidly modulated wideband signal. The low
symbol rate in OFDM makes the use of a guard interval between symbols affordable,
making it possible to eliminate inter symbol interference (ISI).
This is the main advantage of OFDM over FDM. In FDM the data is transmitted
over several carriers which are far apart from each other. The useful data bands are
separated by guard bands to avoid the inter carrier interference (ICI) causing inefficient
spectrum utilization. The usage of guard bands is eliminated in OFDM using the closely
spaced orthogonal subcarriers and the spectrum efficiency is increased. As the subcarriers
are orthogonal to each other they do not interfere with each other.
OFDM is more immune to selective channel fading than single carrier modulation
techniques since it divides the total channel into number of narrowband sub channels,
where the frequency selective fading to broad OFDM signal becomes the flat fading to
the corresponding sub channels. The channel estimation and equalization in OFDM is
much simpler than CDMA or any spread spectrum techniques, since it divides total
bandwidth into number of sub channels. OFDM is widely used in different applications
such as Digital Audio Broadcasting (DAB), Digital Video Broadcasting (DVB), 4G
Mobile Communications, Digital Subscriber Line (DSL) Internet Access, Wireless Local
Area Network (WLAN) called as Wi-Fi based on IEEE 802.11a specifications, power
line networks.

Any communication technique has its own advantages and disadvantages. Like
that OFDM is also having some disadvantages such as high peak to average power ratio
(PAPR), requirement of high synchronism accuracy. Due to high peak to average power
ratio, OFDM system requires high linear power amplifiers otherwise the peaks will be
distorted. There are different techniques to reduce the PAPR such as windowing,
scrambling etc. Even for a small frequency offset the orthogonality among subcarriers is
lost, causing the inter carrier interference, and the information modulated on these
subcarriers is corrupted or lost. So to eliminate the frequency offsets, frequency offset
estimation and correction techniques should be used making the OFDM receiver more
complex.
RESULTS
Code Results
Fig: Original Message
Fig: Recovered message

Fig: Scatter plot

Fig: OFDM transmitted signal
Fig: Received OFDM signal
Simulink Results

Fig: OFDM transmitted signal
Fig: OFDM received signal
Fig: spectrum of OFDM signal

Fig: simulation results for simulink model of OFDM system

CONCLUSION AND FUTURE SCOPE
The importance of multi-carrier communication system has been established in

the present communication era. The merits and demerits of OFDM system along with its
implication are discussed in first chapter. Multi-carrier communication system especially
OFDM has been evolved as one of such potential candidate which are bandwidth
efficient and robust to multipath channel condition (frequency selective fading). The
research activities in OFDM have grown tremendously during last two decades. Due to
its advantageous features like high spectral efficiency, easy equalization and robustness
to frequency selective fading channel, the OFDM has been adopted by many broadband
wireless communication standards like DAB, DVB-T and UWB communication systems.
Besides so many advantageous and favorable features, there exist some major
drawbacks of OFDM which must be resolved for getting all the advantages. Therefore,
for overall improvement in the performance of OFDM system, it is required to handle all
these issues separately. This thesis presents a brief review of major problems of OFDM
system with their existing solutions. The main focus of work was to provide an
appropriate solution to each and every major problem like high PAPR, timing
synchronization, frequency synchronization and ICI reduction. After the review of
different concerns and their solutions, following objectives were set for this study:
Development of a suitable timing offset estimator, Algorithm enumeration for

frequency offset estimation, Establishment of methods for PAPR reduction, and
Induction of new pulse shape in OFDM receiver for ICI reduction . A new technique
which is based on the use of FRFT and chirp signal is proposed for timing offset
estimation in an OFDM system. The proposed method uses chirp signal as a training
sequence and employed the FRFT as a tool to localize the training sequence (chirp) at the
receiver. The comparative study shows the superiority of the proposed estimator in terms
of mean and MSE of timing offset. The MSE of timing offset with proposed method is
found 76% (5 dB SNR) and 63% (8 dB SNR) lower than Awoseyila et.al method, in

HIPERLAN/2 indoor channel-A and in Wi-Max system (strong fading channel)

respectively. However, the betterment in MSE is obtained in the proposed method at the
cost of increased computational complexity, in terms of more complex multiplication
than Awoseyila et.al method. OFDM works on the principle of orthogonality, sub-carriers
are orthogonal to each other. This orthogonality between sub-carriers may be destroyed if
carrier frequency offset arises between them due to Doppler spreading while
transmission. The main task of frequency synchronization in OFDM system is to
estimate and correct this carrier frequency offset at the receiver.

FUTURE SCOPE
In this study, different problems of OFDM system have been considered and
suitable solutions have been provided. As it is an established fact, that research is never
ending process, a new beginning is always waiting. Therefore, following are the works
that may be considered as a future scope in this direction:
 The channel estimation is an area which required a lot of attention and improper
channel estimation degrades the performance of system. In this work, it is
assumed that channel is estimated perfectly. Hence one can evaluate the
performance of proposed work with different channel estimation method.
 The algorithm of timing offset estimation can be extended for channel estimation
in OFDM system.
 The proposed timing offset and frequency offset estimator can be utilized for
MIMO OFDM system.
 The proposed PAPR reduction method can be used with MIMO OFDM system.
 The closed form expression of BER can be derived for OFDM system with
proposed PAPR reduction method.
 The windowing method of ISI reduction can be clubbed with ISI self cancellation
scheme.

REFERENCES
[1] S.S.Ghorpade, S.V.Sankpal,” Behaviour of OFDM System Using MATLAB

Simulation”,(IJITR)..INTERNATIONAL.JOURNAL.OFINNOVATIVE
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International Journal of Emerging Technology and Advanced Engineering, Volume 3,
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[3] Rohith Bodhe, Sathish Narkhede, Shirish Joshi, ”Design of simulink model for ofdm
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[4] Farhan Manzoor, David Linton, Michael Loughlin, “DSP-based Implementation of a

Half-Duplex Orthogonal Frequency Division Multiplexing System” ISSC 2012, NUI
Maynooth, June 28-29, PP. 1-6, 2012.
[5] The MathWorks, Inc., Simulink Getting started guide, Revised for Simulink
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[7] Chien Van Trinh et.al. Implementation of a MIMO-OFDM system based on the TI
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[8] TMS320C6713 DSK technical reference, SPECTRUM DIGITAL,INC.,2004.
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IMPLEMENTATION OF OFDM SYSTEM

Orthogonal frequency division multiplexing (OFDM) is one of the multi-carrier

However, should more subcarriers be required, the modulation, synchronization,

BAPATLA WOMEN’S ENGINEERING COLLEGE 1

inter-carrier interference (ISI) problems. Thanks to these evolutions, modern low-cost

Fig: 1.1 Frequency spectrum of OFDM subcarrier signals

In 1980, Hirosaki introduced an equalization algorithm to suppress both ISI and,

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Fig: 1.2 Magnitude response of a fading channel

In order to deal with frequency-selective fading’s, the transmitted OFDM signals

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1.2 RELATED TECHNOLOGIES

Multi-Carrier Code Division Multiple Access (MC-CDMA), is multi user version

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BAPATLA WOMEN’S ENGINEERING COLLEGE 5

2.1 PRINCIPLE OF OFDM MODEL

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2.1.2 Ofdm modulation

Fig: 2.1 serial-to-parallel conversion

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Fig 2.2: N - Times serial – parallel conversion

2.1.3 Orthogonal Subcarriers

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Fig.2.3: An example of subcarrier waveforms.

2.1.4 Band pass Signaling

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Fig 2.4: OFDM Demodulation

At the receiver, the received signal is down-converted to form a base-band signal.

2.2 MODERN OFDM MODEL

It was a complicated process to achieve N subcarrier oscillators, especially when

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Thanks to IDFT/DFT-based algorithms introduced in 1980. OFDM has become a

Fig 2.5: OFDM model

2.2.1 IDFT and Orthogonality

The orthogonality of discrete-time Fourier bases can be described as follows:

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2.3 GUARD INTERVAL AND CYCLIC PREFIX

In wireless channel, due to multipath propagation and fading the broadening of

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Fig.2.6: Inter Symbol Interference due to Symbol Broadening

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Fig.2.7: Cyclic Prefix is inserted to avoid Inter Symbol Interference

2.3.2 Introduction to Pilot

i. Non data aided channel estimation

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Fig 2.8: GI Eliminates ISI

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Fig.2.9: cyclic prefix

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zero-inserting, and Viterbi decoder components correspondingly. The OFDM modulation

Fig.2.10: simulink model of OFDM.

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The Fourier transform allows us to relate events in time domain to events in

3.1 THE USE OF FFT IN OFDM

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The frequency division multiplexed signal illustrates typical FFT-based OFDM

3.2 SERIAL TO PARALLEL CONVERSION