A

Project Report on
“Noise Cancellation using Adaptive Filter and
performance evaluation of Adaptive
Equalizer”
Submitted in partial fulfillment of the requirement for the award of the degree
of
BACHELOR OF TECHNOLOGY
In
Electronics & Communication Engineering

SUBMITTED BY:

Shubham Mishra(15EC47), Kavita Gangwar(15EC22),

Shubham Gupta(15EC46)
Dept. of Electronics and Communication Engineering,
IET MJP Rohilkhand University Bareilly, India

SUPERVISED BY:
Guide Name
Mr.Janak Kapoor
Assistant Professor
Dept. of Electronics and Communication Engineering,
IET MJP Rohilkhand University Bareilly, India

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 CERTIFICATE
This is to certify that the project entitled “NOISE CANCELLATION USING ADAPTIVE
FILTER AND PERFORMANCE EVALUATION OF ADAPTIVE EQUALIZER
”is a bona fide record work done by KAVITA GANGWAR(15EC22), SHUBHAM
MISHRA(15EC47), SHUBHAM GUPTA(15EC46) during the partial fulfillment of the
requirement for the award of the Degree of Bachelor of Technology.

.
 ACKNOWLEDGEMENT
We deem it a privilege to have been the students of electronics and communication engineering
in IET MJPRU BAREILLY Our heartfelt regards to Mr. Janak Kapoor our project guide who
helped us bring out this project in a fruitful manner with his precious suggestions and rich
experience. We are grateful to Prof. S.K. Tomar, Head of the Department of Electronics &
communication for providing necessary facilities in the department.
 ABSTRACT
This project deals with the study of the various kinds of interferences in a communication
channel viz. Inter symbol Interference, Multipath Interference and Additive Interference. It deals
with the design of an Adaptive Equalizer. The idea of the equalizer is to build (another) filter in the
receiver that counteracts the effect of the channel. In essence, the equalizer must “unscatter” the
impulse response. This can be stated as the goal of designing the equalizer E so that the impulse
response of the combined channel and equalizer CE has a single spike. This can be solved using
different techniques.
The transmission path may also be corrupted by additive interferences such as those caused by
other users. These noise components are usually presumed to be uncorrelated with the source
sequence and they may be broadband or narrowband, in-band or out-of-band relative to the band
limited spectrum of the source signal. Like the multipath channel interference, they cannot be
known to the system designer in advance. The second job of the equalizer is to reject such additive
narrow band interferers by designing appropriate linear notch filters „on-the-fly‟ based on the
received signal. At the same time, it is important that the equalizer not unduly enhance the noise
In this project, we have implemented an „Adaptive Equalizer‟ using Least mean square algorithms
in Matlab. This method(LEAST MEAN SQUARE ALGORITHM) involves minimizing the square
of the error between the received data values and the transmitted values which are achieved via an
adaptive element.
 LIST OF FIGURE

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ABBREVIATION USED
CONTENT
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CHAPTER 1: INTRODUCTION
1.1 GENERAL
Our requirement is to design a filter which minimizes the
effect of the noise at the filter output by using some statistical
criterion. An optimal solution to this problem is minimizing the
mean square value of the error signal. This approach of
minimizing error signal to the stationary input data is known as
Wiener Filter. But this wiener filter is not useful for non-
stationary data.
The design of wiener filter also requires prior information
on the signal which cannot be possible at every time. So we
need to find the solution for the filter designing which do not
require any relevant signal information. Thus we go for
Adaptive Filter which meets our requirement.

1.2 ADAPTIVE FILTER

The adaptive filter can be defined as, a filter which itself
regulate its switch function according to some optimizing
parameter. The filter coefficients are up-to-date at every
generation until they converge to long-established signal.
Centered on the precise signals bought, it makes an attempt to
find the most excellent filter design. In a non -stationary
atmosphere, the filter is anticipated to track time versions and
range its filter coefficients, accordingly in a stationary
environment, the filter is anticipated to converge, to the Wiener
filter.
The operation of adaptive filter involves two processes:
1) Filtering
2) Weight adjustment procedure

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General description for the figure are given below, x(n)-Input signal
plus Noise, d(n)-Desired signal, e(n)-Error signal, y(n)output signal.

Figure 1.2 Basic block diagram of an adaptive filter used for noise cancellation

There are some types of adaptive algorithms like

1) Least Mean Square (LMS) algorithm
2) Recursive Least Square (RLS) algorithm

1.2.1 Least Mean Square (LMS):

The LMS algorithm is a linear adaptive filtering algorithm that

consists of two basic processes. The first process is the filtering
process, which involves computations of the transversal filtering
output of a produced by a set of tap inputs and estimation error by
comparing this output to a desired response. Next, the adaptive
filtering will adjust automatically the filter tap weights in
accordance with the estimation error.
Consistent with this LMS algorithm the burden replace
equation is given in equation
W(n + 1) = W(n) + 2 * p* E(n) *X(n)
Where, p = Step Size,
W(n)=Weight update, X(n) = Input signal, E(n)= Error signal

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1.2.1 Flow Chart For LMS adaptive filter.

Figure 1.2.1 Flow chart of LMS adaptive filter

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1.3 SIMULATION RESULTS

So we plotted our desired signal,output signal and voice signal

with by using MATLAB as shown below:-

Figure 1.3.1 Obtained Simulation results using MATLAB showing (a)

voice signal without noise (b) voice signal with noise (c) noisy voice after
filtering. Parameters (i) Order of Filter N=20 (ii) Convergence Factor
µ=0.01

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Figure 1.3.2 Obtained Simulation results using MATLAB showing (a)

voice signal without noise (b) voice signal with noise (c) noisy voice after
filtering. Parameters (i) Order of Filter N=20 (ii) Convergence Factor
µ=0.03

Figure 1.3.3 Obtained Simulation results using MATLAB showing (a)

voice signal without noise (b) voice signal with noise (c) noisy voice after
filtering. Parameters (i) Order of Filter N=12 (ii) Convergence Factor
µ=0.03

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1.4 AIM AND OBJECTIVE: performance evaluation of

Adaptive Equalizer using LMS algorithm

1.5 PROBLEM STATEMENT: When all is well in receiver there is

no interaction between successive symbol each symbol arrives and is
decoded independently of all others. But when symbols interact, when
the waveform of one symbol corrupts the value of a nearby symbol,
then the received signal becomes distorted. It is difficult to decipher the
message from such a received signal. This impairment is called “inter-
symbol interference”.

When there is no inter-symbol interference (from a multipath channel,

from imperfect pulse shaping, or from imperfect timing), the impulse
response of the system from the source to the recovered message has a
single nonzero term. The amplitude of this single “spike” depends on the
transmission losses, and the delay is determined by the transmission
time. When there is inter-symbol interference caused by the channel, this
single spike is “scattered”, duplicated once for each path in the channel.
The number of nonzero terms in the impulse response increases. The
channel can be modeled as a finite-impulse-response, linear filter C, and
the delay spread is the total time interval during which reflections with
significant energy arrive. The idea of the equalizer is to build (another)
filter in the receiver that counteracts the effect of the channel. In essence,
the equalizer must “unscatter” the impulse response. This can be stated
as the goal of designing the equalizer E so that the impulse response of
the combined channel and equalizer CE has a single spike. This can be
cast as an optimization problem, and can be solved using different
techniques.

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CHAPTER 2:LITERATURE REVIEW

2.1 GENERAL

2.1.1 Fundamentals of equalization

Inter symbol interference (ISI) has been recognized as a major

obstacle to high speed data transmission. For high speed data
transmission, we have no choice but to use some kind of
equalization technique.
Equalization is used to combat inter symbol interference, as the
fading channels are random as well as time varying equalizers must
track the time varying characteristics of the mobile channel and thus
are called adaptive equalizers.
Blind equalization wherein it doesn’t need a training sequence.It
uses some property of the signal may be constant envelop or a
constant amplitude. So as to guess the distortion being put in by the
channel and undo those effects but the more common once are
adaptive equalizers which work on the concept of a training
sequence.

2.2 TYPES OF EQUALIZER:-

Depending on the time natureequalizers can be grouped as preset or
adaptive equalizers.

2.2.1 Preset equalizers:

assume that the channel is time invariant
and try to find H(f) and design equalizer depending on H(f).
The examples of these ADAPTIVE EQUALIZERS are zer o
forcing equalizer, minimum mean square error equalizer,
and desicionfeedback equalizer.

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2.2.2 ADAPTIVE EQUALIZER:

An adaptive equalizer is an equalization filter that automatically
adapts to time-varying properties of the communication channel. It
can be implemented to perform tap-weight adjustments periodically
or continually. Periodic adjustments are accomplished by periodically
transmitting a preamble or short training sequence of digital data
known by the receiver. Continual adjustment are accomplished by
replacing the known training sequence with a sequence of data
symbols estimated from the equalizer output and treated as known
data. When performed continually and automatically in this way, the
adaptive procedure is referred to as decision directed.
If the probability of error exceeds one percent, the decision directed
equalizer might not converge. A common solution to this problem is
to initialize the equalizer with an alternate
process, such as a preamble to provide good channel-error
performance, and then switch to decision-directed mode.

2.3 OPERATION OF ADAPTIVE EQUALIZER:

The two operating modes of an adaptive equalizers are:
1.Training mode.
2. Tracking mode.
The training mode proceeds the tracking mode.

2.3.1 Training mode of adaptive equalization

Initially a known fixed length training sequence is sent by the

transmitter so that the receiver’s equalizer may average to a proper
setting.
The training sequence is usually a pseudo random sequence or a
fixed known prescribed bit pattern usually put in the standard.
Immediately following the training sequence the user data is sent.

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The training sequence in an adaptive equalizer is designed

to permit and equalizer at the receiver to acquire the
proper filter coefficients in the worst possible channel
conditions.
Therefore when training sequence is finished, the filter
coefficients are near optimal
An adaptive equalizer at the receiver uses a LMS
algorithms to evaluate the channel and estimate filter
coefficients to compensate for the channel.

2.3.2Tracking mode

When the data of the users are received, the adaptive

algorithms of the equalizer tracks the changing channel.
As a result of this the adaptive equalizer continuously
changes the filter characteristics over time.
2.4 BLOCK DIAGRAM OF AN ADAPTIVE
EQUALIZER:

Figure 2. 4Block Diagram Of Adaptive Equalizer

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We have the original baseband message x(t), it passes

through a modulator then through the transmitter over to
the radio channel. Here noise and fading effects get added
leading to a very distorted received signal.
Here it goes through the RF receiver front end, passes through
the IF stage and then detector in match filter. So equalizer
usually works either at the IF stage or at the baseband. So here
is equalizer, decision maker and error which kind of tracks
and then finally we have the reconstructed message data d(t).
In the training section this d(t) and x(t) should be equal.

2.2.3 STRUCTURE OF AN EQUALIZER-FIR

FILTER:

A finite impulse response (FIR) filter is a type of a digital filter.

The impulse response is finite because it settles to zero in a finite
number of sample intervals. This is in contrast to infinite impulse
response (IIR) filters, which have internal feedback and may
continue to respond indefinitely. The impulse response of an Nth-
order FIR filter lasts for N+1 samples, and then dies to zero.

We start the discussion by stating the difference equation which defines

how the input signal is related to the output signal:

where:
x[n] is the input signal,
y[n] is the output signal, and
bi are the filter coefficients.
N is known as the filter order; anNth-order filter has (N + 1) terms on the right-
Hand side are commonly referred to as taps.
The previous equation can also be expressed as a convolution of filter coefficients
And the input signal

In general, an accurate digital model for a channel depends on many things: the
underlying analog channel, the pulse shaping used, and the timing of the sampling
process. At first glance, this seems like it might make designing an equalizer for such a
channel almost impossible. But there is good news. No matter what timing instants are
chosen, no matter what pulse shape is used, and no matter what the underlying analog
channel may be (as long as it is linear), there is a FIR linear representation of the form
(1.1) that closely models its behavior.
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Advantages of FIR Filters (compared to IIR filters:

1.They can easily be designed to be "linear phase" (and usually are). Put simply, linear-
phase filters delay the input signal, but don‟t distort its phase.

2. They are simple to implement. On most DSP microprocessors, the FIR calculation can
be done by looping a single instruction.

3. They are suited to multi-rate applications. By multi-rate, we mean either "decimation"

(reducing the sampling rate), "interpolation" (increasing the sampling rate), or both.
Whether decimating or interpolating, the use of FIR filters allows some of the
calculations to be omitted, thus providing an important computational efficiency. In
contrast, if IIR filters are used, each output must be individually calculated, even if it that
output will discarded (so the feedback will be incorporated into the filter).

4. They have desirable numeric properties. In practice, all DSP filters must be
implemented using "finite-precision" arithmetic, that is, a limited number of bits. The use
of finite-precision arithmetic in IIR filters can cause significant problems due to the use
of feedback, but FIR filters have no feedback, so they can usually be implemented using
fewer bits, and the designer has fewer practical problems to solve related to non-ideal
arithmetic.

5. They can be implemented using fractional arithmetic. Unlike IIR filters, it is always
possible to implement a FIR filter using coefficients with magnitude of less than 1.0.
(The overall gain of the FIR filter can be adjusted at its output, if desired.) This is an
important consideration when using fixed-point DSP's, because it makes the
implementation much simpler.

Disadvantages of FIR Filters (compared to IIR filters):

Compared to IIR filters, FIR filters sometimes have the disadvantage that they require
more memory and/or calculation to achieve a given filter response characteristic. Also,
certain responses are not practical to implement with FIR filters.

2.3 BASIC MODEL OF A COMMUNICATION SYSTEM:

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Figure 2. 3 basic block diagram of communication system

The signal path of a baseband digital communication system is shown in fig 1.1, which
emphasizes the role of the equalizer in trying to counteract the effects of the multipath
channel and the additive interference.
In the figure above, a digital input is passed thrrough an analog channel after pulse
shaping. Pulse shaping is the process of changing the waveform of transmitted pulses. Its
purpose is to make the transmitted signal suit better to the communication channel by
limiting the effective bandwidth of the transmission. By filtering the transmitted pulses
this way, the intersymbol interference caused by the channel can be kept in control. Then
after passing through the analog channel noise and interferers are added to the signal in
order to account for the actual noise and interferences which a signal encounters while
passing through a channel.
The analog signal is sampled and then it is passed through a linear digital equalizer. The
baseband linear (digital) equalizer is intended to ( automatically) cancel unwanted effects
of the channel and to cancel certain kinds of additive interferences.
All of the inner parts of the system are assumed to operate precisely. Thus the up and
down conversion, the timing recovery and the carrier synchronization are assumed to be
flawless and unchanging. Modeling the channel as a time invariant FIR filter, the next
section focuses on the task of selecting the coefficients in the block labeled “linear digital
equalizer”, with the goal of removing the inter symbol interference and attenuating the
additive interferences. These coefficients are to be chosen based on the sampled received
signal sequence and (possibly) knowledge of a prearranged “training sequence”. While
the channel may actually be timevarying, the variations are often much slower than the
data rate, and the channel can be viewed as (effectively) time-invariant over small time
scales.

2.4 MULTIPATH INTERFERENCE

The villains of communication are multipath and other additive interferers. The distortion
caused by an analog wireless channel can be thought of as a combination of scaled and
delayed reflections of the original transmitted signal. These reflections occur when there
are different paths from the transmitting antenna to the receiving antenna. The strength of

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the reflections depends on the physical properties of the reflecting objects, while the
delay of the reflections is primarily determined by the length of the transmission path.

Let u(t) be the transmitted signal. If N delays are represented by ∆1, ∆2, ∆3, ∆4…∆N and
the strength of the reflections is α1, α2, α3, α4,…αN then the received signal y(t) is

Where η(t) represents additive interferences. This model of the transmission channel has
the form of a finite impulse response filter, and the total length of time ∆N - ∆1 over
which the impulse response is nonzero is called the delay spread of the physical medium.
This transmission channel is typically modeled digitally assuming a fixed sampling
period Ts.Thus (1.2) is approximated by

In order for the model (1.3) to closely represent the system (1.2), the total time over
which the impulse response is nonzero( the time ηTs) must be at least as large as the
maximum delay ∆N .Since the delay is not a function of the symbol period Ts, smaller Ts
require more terms in the filter, i.e., larger n.

For example, consider a sampling interval of Ts = 40 nanoseconds (i.e. a transmission

rate of 25 MHz). A delay spread of approximately 4 microseconds would correspond to
one hundred taps in the model (1.3). Thus at ant time instant, the received signal would
be a combination of (up to) one hundred data values. If Ts were increased to 0.4
microseconds (i.e 2.5 MHz), only ten terms would be needed, and there would only be
interference with the ten nearest data values. If Tswere larger than 4 microseconds (i.e.
0.25 MHz),only one term would be needed in the discrete-time impulse response. In this
case, adjacent sampled symbols would not interfere. Such finite duration response models
can also be used to represent the frequency-selective dynamics that occur in the wired
local end –loop in telephony, and other (approximately) linear, finite-delay spread
channels.
The design objective of the equalizer is to undo the effects of the channel and to remove
the interference. Conceptually, the equalizer attempts to build a system that is a “delayed
inverse” of (1.3), removing the inter symbol interference while simultaneously rejecting
additive interferers uncorrelated to the source. If the interference η(kTs) is unstructured
(for instance white noise) then there is little that a linear equalizer can do to improve it.
But when the interference is highly structured (such as narrow band interference from
another user) then the linear filter can often notch out the offending frequencies.

CHAPTER 3
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3. ALGORITHMS FOR THE IMPLEMENTATION OF ADAPTIVE

EQUALIZER IN MATLAB:

3.1 AN ADAPTIVE APPROACH TO THE TRAINED

EQUALIZATION:

The block oriented design of the previous section requires substantial computation even
when the system delay is known since it requires the calculating the inverse of an (n+1) X
(n+1) matrix, when n is the largest delay in the FIR linear equalizer. This section
considers using an adaptive element to minimize the average of the squared error.

JLMS = ½ avg{e2[k]}
Observe that JLMS is a function of all the equalizer coefficients fi since

which combines (1.10) with (1.4) and where r[k] is the received signal baseband after
sampling. An algorithm for the minimization of JLMS with respect to the ith equalizer
coefficient fi is

To create an algorithm that can be easily implemented, it is necessary to evaluate this

derivative with respect to the parameter of interest. This is

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Typically, the averaging operation is suppressed since the iteration with small step-size
itself has a low pass (averaging) behavior. This result is commonly called the Least Mean
Squares (LMS) algorithm for direct linear equalizer impulse response coefficient
adaptation.

The adaptive equalization scheme is illustrated in the Figure:

Figure 3. 1Trained Adaptive Linear Equalizer

When all goes well, the recursive algorithm converges to the vicinity of the block least
squares answer for the particular used in forming the delayed recovery error. As long as
µ is nonzero,
if the underlying composition of the received signal changes so that the error increases
and the desired equalizer changes, then fi react accordingly. It is this tracking ability that
earns it the label adaptive.

CHAPTER 4
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RESULT AND DISSCUSSION

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