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ABSTRACT
INTRODUCTION
CHANNEL ESTIMATION IN OFDM
OFDM & BLOCK DIAGRAM OF OFDM
WHY OFDM?
FEATURES OF OFDM
CHANNEL EQUALISATION
INTER SYMBOL INTERFERENCE
MATLAB
RESULTS AND ANALYSIS
CONCLUSION
REFERENCES
ABSTRACT
Nowadays the scope of wireless communication has increased to a great extent because of the use of multicarrier modulation
techniques, among which OFDM is one of the most popular technique.
OFDM efficiently deals with multipath fading, channel delay spread, enhancement of channel capacity, modification of modulation
density and robustness of narrowband interference.
OFDM communication system consists of channel model through which data symbols are transmitted to the receiver. This channel
model produces line of sight communication and also various reflections due to which multipath effect come to picture.
To minimize the multipath effect and noise introduced by the channel, we go for Channel Estimation.
In OFDM systems, at the transmitter& receiver DAC, ADC are used respectively which never have exactly the same sampling
period. Due to this, intercarrier interference and the slow shift of the symbol timing point occurs and so Orthogonality is lost. This
results in need of Synchronization and Channel Equalization.
In this project, implementation of OFDM communication system with channel estimation and the bit error rate (BER) of OFDM
system with and without channel estimation is observed with different modulation schemes such as QPSK,16QAM,64QAM using
MATLAB
INTRODUCTION
Orthogonal frequency division multiplexing(OFDM) is a form of multicarrier modulation. An OFDM signal consists of a number of closely spaced
modulated carriers. When modulation of any form - voice, data, etc. is applied to a carrier, then sidebands spread out either side. It is necessary for a
receiver to be able to receive the whole signal to be able to successfully demodulate the data. As a result when signals are transmitted close to one
another they must be spaced so that the receiver can separate them using a filter and there must be a guard band between them. This is not the case
with OFDM. Although the sidebands from each carrier overlap, they can still be received without the interference that might be expected because
they are orthogonal to each another. This is achieved by having the carrier spacing equal to the reciprocal of the symbol period.
CHANNEL ESTIMATION IN OFDM
Channel Estimation is basically the estimation of channel effect on the signal I.e,the amplitude and phase shift
caused by the channel
Channel estimation plays an important part in an OFDM system. It is used for increasing the capacity of orthogonal
frequency division multiple access (OFDMA) systems by improving the system performance in terms of bit
error rate.
Here we do the pilot symbol assisted channel estimation.
Pilot subcarriers transmit with a known data sequence. This information is used to determine the difference, or error,
between an ideal signal and the actual received signal.
This analysis plays a key role in the transmission as it is useful in reducing the BER.
Fig: Communication system model
BLOCK DIAGRAM FOR LS ESTIMATION
TECHNIQUE
Block diagram for LS estimation There are mainly two types of estimators used in the channel estimation called minimum
mean square error (MMSE) and least squares (LS) channel estimators.
The MMSE estimator has good performance but high complexity. The LS estimator has low complexity, but its performance is
not as good as that of the MMSE estimator.
Modifications to both MMSE and LS estimators that use the assumption of a finite length impulse response are presented so
that the performance of the estimators can be increased
FLOWCHART
Flow chart describing the sequence of steps followed in obtaining
the output signal observed under ls estimation technique using
various modulation schemes
OFDM, Orthogonal Frequency Division Multiplexing is a form of signal waveform or modulation that provides
some significant advantages for data links.
It is a particular case of multicarrier transmission in which, higher data rates are achieved using carriers that are
densely packed.
Accordingly, OFDM, Orthogonal Frequency Division Multiplexing is used for many of the latest wide bandwidth
and high data rate wireless systems including Wi-Fi, cellular telecommunications and many more
WHY OFDM?
The OFDM is a very efficient modulation technique that can achieve very high throughput by transmitting on a great number of
carriers simultaneously.
OFDM, like FDM, separates the channel bandwidth into multiple narrow-band subcarriers to carry the information. To prevent
adjacent carrier interference (ACI),
Traditional FDM systems require small gaps or guard bands between the carriers where no information can be transmitted. This
results in a waste of spectrum. To solve this problem, OFDM uses special subcarriers that are all orthogonal to each other.
This process not only permits the removal of the guard bands, but since the subcarriers are completely unrelated, they can even
overlap each other. This is why OFDM is so bandwidth efficient.
to prevent inter-symbol interference (ISI) caused by the propagation channel, OFDM systems insert a cyclic prefix (CP) before
each symbol to be transmitted. To preserve orthogonality, the end of the current symbol is transmitted before each symbol.
BLOCK DIAGRAM OF OFDM:-
FEATURES OF OFDM:-
The OFDM scheme differs from traditional FDM in the following interrelated ways:
Multiple carriers (called subcarriers) carry the information stream
The subcarriers are orthogonal to each other.
A guard interval is added to each symbol to minimize the channel delay spread and
intersymbol interference(ISI).
CHANNEL EQUALIZATION
The term channel equalization stems from the requirement that the amplitude response of the channel must be flat,
that is, the same for all frequencies of the channel.
If the amplitude response is not flat, it must be equalized using an appropriate device at the receiver, which is called
an equalizer.
Types of equalizers available are
Linear equalizer:
• MMSE equalizer : MMSE stands for minimum mean square error,designs the filter to minimize E[|e|2], where e is
the error signal, which is the filter output minus the transmitted signal.
• Zero Forcing Equalizer refers to a form of linear equalization algorithm used in communication systems which
applies the inverse of the frequency response of the channel
Decision feedback equalizer: augments a linear equalizer by adding a filtered version of previous
symbol estimates to the original filter output.
Blind equalization: it is a digital signal processing technique in which the transmitted signal is inferred
(equalized) from the received signal, while making use only of the transmitted signal statistics. Hence,
the use of the word blind in the name.
Adaptive equalizer: it is an equalizer that automatically adapts to time-varying properties of
the communication channel. It is frequently used with coherent modulations such as phase shift keying,
mitigating the effects of multipath propagation and Doppler spreading
Viterbi equalizer: FThe Viterbi algorithm is a dynamic programming algorithm for obtaining
the maximum a posteriori probability estimate of the most likely sequence of hidden states—called
the Viterbi path—that results in a sequence of observed events
BCJR equalizer: uses the BCJR algorithm (also called the Forward-backward algorithm) to find
the maximum a posteriori (MAP) solution. Its goal is to minimize the probability that a given bit was
incorrectly estimated.
Turbo equalizer: applies turbo decoding while treating the channel as a convolutional code.
Among all these we are employing our OFDM system with the Adaptive Equalizer
Inter symbol interference
To compensate the time dispersion of the channel and to reduce the ISI, several techniques are available. But, to
adequately characterize the effects of all channel types, channel equalizers are adopted.
When there is no interaction between successive symbols at the receiver, then each symbol arrives independently
and decoded independently by the users.
But when symbols interact, the waveform of one symbol corrupts the value of a nearby symbol, then the received
signal gets distorted and it is difficult to interpret the message from such a received signal. This overlapping of the
nearby symbols is called “Inter Symbol Interference” (ISI).
It occurs because of multipath channel, imperfect pulse shaping, and imperfect timing
Fig. Inter Symbol Interference.
When there is no ISI the impulse response of the system has a single spike.
The amplitude of this single “spike” depends on the transmission losses, and the delay is determined by the transmission
time.
When there is ISI caused by the multipath channel, this single spike is “scattered”, duplicated once for each path in the
channel. Then, the number of nonzero terms in the impulse response increases.
The channel equalizer reconstructs or estimates the corrupted data sequence from a set of received symbols.
The adaptive equalizer concept was introduced by R. W. Lucky on the topic “Peak Distortion Criterion” in
1965
Adaptive equalizer self-adjusts its transfer function according to an optimizing algorithm.
These optimizing algorithms update the equalizer coefficients and track the channel changes.
Block diagram of adaptive transversal filter employing LMS algorithm is shown in above Fig.
Since the output sequence has tried to adapt to the input reference sequence, this type of equalizer is
called the adaptive equalizer.
Two features, simple to design and very effective in performance have made this algorithm highly
popular in various applications.
LMS ALGORITHM:
To adjust the system parameters which are changing adaptively, the linear transversal filtering technique Least
Mean Squares (LMS) is used.
The filter is adapted based on the error at the current time. (Widrow and Hoff, 1960) proposed LMS algorithm, to obtain
optimum filter weights
The main principle of this algorithm is to regulate the equalizer tap coefficients with the reference to the required response,
however, it converges slowly.
LMS algorithm takes the mean square of the error function and differentiates it at number of iterations. For every iteration, the
error gets reduced, and finally become zero.
Adjusts the weight w(n) of the filter.
Slow in convergence and sensitive to variations in step size parameter. Requires number of iterations equals to dimensionally
of the input.
ALGORITHM:
The LMS algorithm is a linear adaptive algorithm and it performs two fundamental processes on the
signal.
1. Filtering process : This involves, computing the output of a linear filter in response to the input
signal and generating an estimation error by comparing filter output with the desired response.
a) Filter output:
b) Estimation of error or error signal ...(i)
…(ii)
where e(n) is the estimated error signal, y(n) is filter output, d(n) is the desired response at time n. Above
equations show the calculation of estimation error e(n), decided on the present estimate of the tap weight
vector, ŵ (n)
The weights of the filter are adjusted using the error which is estimated in filtering process given in
Eq.(ii)
Where ' ' is the step size, is the estimate of tap weight vector at time (n+1), is the tap
weight vector at time (n)
If ' ' is too small the algorithm converges slowly and may not be able to track changing conditions.
Where λmax is the largest Eigen value of the autocorrelation matrix for input signal. The performance of
the algorithm is influenced by the value of
The convergence speed is faster with greater value of , but when its value is too large; oscillation will
occur during the convergence. However, the decrease of value will reduce the convergence speed and
tracking speed.
It is important to take into consideration that x(n) e*(n) term shows adjustment which is applied to the
present estimate of the tap weight vector, ŵ (n)
In LMS algorithm, the change of individual tap weights for each iteration are controlled by the obtained
error value e(n).
LMS algorithm is most popular because of its convergence speed, but selection of step size is very
important in the case of success of the algorithm
MATLAB
PADDING: By default pad array adds padding before the first element and after the last element of
each dimension. B = padarray( A , padsize , padval ) pads array A where padval specifies a constant
value to use for padded elements or a method to replicate array elements.
RESHAPE: The reshape function changes the size and shape of an array. For example, reshape a 3-by-
4 matrix to a 2-by-6 matrix. As long as the number of elements in each shape are the same, you can
reshape them into an array with any number of dimensions. Using the elements from A , create a 2-by-
2-by-3 multidimensional array
-1
BER in AWGN considering QPSK
10
-2
10
-3
10
BER
-4
10
-5
10 simulated BER
Theoretical BER
-6
10
0 1 2 3 4 5 6 7 8 9 10
Eb/No(dB)
Fig. BER analysis of AWGN channel considering QPSK Fig.BER analysis of FSF channel using LMS equalization and
modulation considering QPSK modulation
CONSTELLATION DIAGRAMS ON TRANSMITTER
SIDE AND RECEIVER SIDE OF
QPSK MODULATION FOR BLACK AND WHITE
IMAGE
Fig : channel equalization effect of QAM on black and white image of QPSK for 50db SNR
COMPARISON OF CHANNEL
ESTIMATION WITH OFDM IN MATLAB USING
PHASE SHIFT KEYING (QPSK) TECHNIQUE:
Fig: Transmitter constellation, Receiver constellation and BER of QPSK modulation for 50db SNR
Fig: Transmitter constellation, Receiver constellation and BER of QPSK modulation
for 10db SNR
Fig: Transmitter constellation, Receiver constellation and BER of QPSK modulation
for 20db SNR
Fig: Transmitter constellation, Receiver constellation and BER of QPSK modulation
for 30db SNR
Fig: Transmitter constellation, Receiver constellation and BER of QPSK modulation
for 40db SNR
Above results are obtained by taking a black and white image and the corresponding results are obtained
for 50dB SNR using Qpsk modulation scheme
It is repeated for various other values of SNR like 10dB,20dB,30dB,40dB and also repeated for another
modulation schemes also such as 16QAM&64QAM
It can be seen that the BER decreases with increase in SNR, and also BER is better for QPSK than
16QAM and 64QAM
And also we can observe that the quality of received image is improved while using OFDM than the
one we got without OFDM
CONSTELLATION DIAGRAMS ON TRANSMITTER
SIDE &RECEIVER SIDE OF QPSK FOR COLOUR
IMAGE
q It is repeated for various other values of SNR like 10dB,20dB,30dB,40dB and also repeated for another
modulation schemes also such as 16QAM&64QAM
q It can be seen that the BER decreases with increase in SNR,and also BER is better for QPSK than
16QAM and 64QAM
q And also we can observe that the quality of received image is improved while using OFDM than the
one we got without OFDM
CONCLUSION
In this project we have implemented the channel equalization technique with and without OFDM using
QPSK,16QAM,64QAM modulation schemes using MATLAB.
We have observed the effect of channel equalization on image and correspondingly the constellation diagrams have
also been observed.
From this project it is concluded that the Bit Error Rate(BER) reduces for QPSK Than 16QAM,64QAM with the
usage of OFDM technique rather than without OFDM transmission.
REFERENCES
[1]An Experimental study on Channel estimation and synchronization to reduce error rate in OFDM using GNU
radio International conference on Information and communication Technologies(ICICT 2017)
[2] OFDM: Today and in the Future of Next Generation Wireless Communications by Hernan F. Arrano, Cesar
A. Azurdia-Meza, Conference Paper · September 2016
[3] OFDM Modulation Technique &its Applications, International conference on innovations in computing(ICIC
2017).
[4] Rappaport, T.S., Wireless Communications Principles and Practice, IEEE Press, New York, Prentice Hall, pp.
169-177, 1996.
[5] Nisha Solanki. “Digital Audio Broadcasting”, Sarjan SOCET Journal, vol 1, issue1, Sept 2012.
[6] L. Thibault, Minh Thien Le, “Performance evaluation of COFDM for digital audio broadcasting. I.
Parametric study”, IEEE Transactions on Broadcasting, vol: 43, Mar 1997.
[7] M. Alard, R. Halbert, B. Le Floch, D. Pommier, "A new system of sound broadcasting to mobile
receivers", Eurocon Conference 1988.
[8] Beena R. Ballal1, Ankit Chadha2, Neha Satam, Orthogonal Frequency Division Multiplexing and its
Applications, International Journal of Science and Research (IJSR), India Online ISSN: 2319-7064.
[11] B. Le Floch, M. Alard, C. Berrou, "Coded Orthogonal Frequency Division Multiplex", Proceedings of the
IEEE, vol. 83, no. 6, pp. 982-996, June 1995.
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