Mehta Vidhi (4102)

Mistry Nitisha (4105)

Patel Dhruvi (4118)
What is an Adaptive filter?
An Adaptive filter is a filter that self-adjusts
its transfer function according to an
optimizing algorithms.
Most adaptive filters are digital filters that
perform digital signal processing and adjust
their performance based on input signal.
And are now used in device such as mobile
phone ,camcorders, digital cameras etc.
 Block Diagram Of Adaptive
Filter

Where
x(n) is the input signal to a linear filter at time n
y(n) is the corresponding output signal
d(n) is an additional input signal to the adaptive filter
e(n) is the error signal that denotes the difference
between d(n) and y(n)
 Cont…
The output of programmable of variable
digital filter is subtracted from a desired
signal that is d(n) to produce an error
sequence e(n) which is used in combination
with elements of the input sequence x(n), to
update the filter coefficients which is to be
minimized.
 Discussion of Block Diagram
 To start the discussion of the block diagram we take the
following assumptions:

 The input signal is the sum of a desired signal d(n) and

interfering noise v(n)
 x(n) = d(n) + v(n)

 The variable filter has a Finite Impulse Response (FIR)

structure. For such structures the impulse response is
equal to the filter coefficients. The coefficients for a
filter of order p are defined as

 Wn = [Wn(0), Wn(1),…,Wn(p)]T

 The error signal or cost function is the difference

between the desired and the estimated signal

 e(n) = d(n) – d’(n)

Cont…
The variable filter estimates the desired signal by convolving the input
signal with the impulse response. In vector notation this is expressed
as

Where X(n) is an input signal vector.

Cont..
The variable filter updates the filter
coefficients at every time instant

Where ∆Wn is a correction factor for the filter coefficients. The

adaptive algorithm generates this correction factor based on
the input and error signals.
Cont…
• There are two different algorithm to update
the co-efficient of filter:---

1.Least Mean Square(LMS) Algorithm.

2.Recursive least squares (RLS) Algorithm.
1. Least mean
squares(LMS) filter
Why LMS ???
In our system, the music that is being played by the
computer is a known output that is being fed into the
room, which has an unknown response, and picked up by
the microphone. Since we have the signal both before
and after the influence of the room, this is a candidate for
an adaptive filter. The adaptive filter can be used to try to
estimate the room’s response based on the music being
outputted and the input to the microphone. Once an
estimate is available, we can use it to remove the
interference of the music from the input to the
microphone. .
LMS Overview
One of the most popular adaptive algorithms used today is
the Least Mean Squares (LMS) algorithm. Essentially, this
algorithm attempts to minimize the error that occurs between
the detected (or desired signal), d[n], and the estimated
value of the signal, y[n] and find the filter coefficients that
relate to producing the least mean squares of the error
signal . This estimated signal is created by taking the original
input, x[n] , and running it through an approximation of the
unknown channel. Basically:

y[n]= h(n)x T(n)

Cont..

x[n] is the input to the filter,

v[n] is the interference in the room,
h[n] is the impulse response(weights),
e[n] is the error in this modeling.
e(n)=d(n)-y’(n)
Cont..
Using the error, we can approximate the next set of weights
as follows:
h’(n+1) = h’(n) + μe*(n)x(n)
where µ is a constant.

Choosing µ:--
The choice of µ is an important one as it greatly
affects how the system will perform. If the value is too
small, then the adaptive filter will not adapt fast enough.
If it is too large, the system will not converge at all, and the
value weights will actually diverge.
 2. Recursive least
squares (RLS) Filter.
Recursive least squares (RLS) algorithm is used in
adaptive filters to find the filter coefficients that relate to
recursively producing the least squares (minimum of the sum
of the absolute squared) of the error signal. This is contrast
to other algorithms that aim to reduce the mean square error
. The difference is that RLS filters are dependent on the
signals themselves, whereas MSE filters are dependent on
their statistics.
In the RLS method, the error considered is the total error
from the beginning to the current data point.
Cont..
The idea behind RLS filters is to minimize a cost
function C by appropriately selecting the filter
coefficients wn, updating the filter as new data
arrives. The error signal e(n) and desired signal
d(n) are defined in the negative feedback diagram
below:
Cont..
The RLS algorithm for a p-th order RLS filter can be summarized
as
Parameters: p = filter order
λ = forgetting factor
δ = value to initialize P(0)
Initialization: wn=0
P(0)= δ -1I where I is the (p + 1)-by-(p + 1) identity matrix
Computation: For n=0,1,2,….
x(n)=[x(n),x(n-1),…,x(n-p)]T
a(n) = d(n) - w(n-1) T x(n) [Prior error]
g(n) = P(n)x*(n) [Gain vector]
P(n)= λ-1P(n-1)- g(n) x T(n) λ-1P(n-1)
w(n) = w(n-1) + a(n)g(n)
Cont..
Choosing λ:--
As λ approaches zero, the past errors play a
smaller role in the total. This makes the filter
more sensitive to recent samples, which means
more fluctuations in the filter co-efficients.

Since 0 ≤λ< 1, applying the factor is equivalent to

weighting the older error. When λ = 1, all previous
error is considered of equal weight in the total error.
LMS Algorithm V/s RLS
Algorithm
1. Note that the signal paths and identifications are the
same whether the filter uses RLS or LMS. The
difference lies in the adapting portion.

2. While minimizing the error using LMS algorithm, it

considers only the current error value. In the RLS
method, the error considered is the total error from the
beginning to the current data point.
 Cont..
3. The LMS algorithm requires approximately 20M
iterations to converge in mean square, where M is the
number of tap coefficients contained in the tapped-delay-
line filter. On the other band, the RLS algorithm converges
in mean square within less than 2M iterations. The rate of
convergence of the RLS algorithm is therefore, in general,
faster than that of the LMS algorithm by an order of
magnitude.

4. In contrast to the least mean squares algorithm, from

which it can be derived, the RLS adaptive algorithm
minimizes the total squared error between the desired
signal and the output from the unknown system.
Cont..
5. The superior performance of the RLS algorithm
compared to the LMS algorithm, however, is attained at
the expense of a large increase in computational
complexity. i.e. if we want something best than we use
complex function in the same way here RLS use
complex computation

6. One interesting input option that applies to RLS

algorithms is not present in the LMS processes — a
forgetting factor, λ, that determines how the algorithm
treats past data input to the algorithm.
Applications of Adaptive filter:-
1) Channel equalization and Channel-
identification
2) Noise cancellation
3) Signal prediction
4) Adaptive Feedback Cancellation
• It’s like when we compose
music,how channel is equalized so it is an
electronic development for combining and
changing the level of audio channel.

• Example:
In TV you see musician changing
signals that all are done on sound board.
 2). Noise cancellation
• It is also known as active noise channel or active
noise reduction.
• It is a method used for reducing unwanted
sound.
• Antinoise is used to reduce noise at the working
environment with ear plugs.
• Bigger noise cancellation systems are used for
ship engines
Example:
When DJ plays music he always wear head
phones to avoid unwanted sound.
 Real Time Example of Noise Cancellation
Application(Pilot):-

• The purpose of adaptive noise cancellation is to

improve the signal-to-noise ratio (SNR) of a signal by
removing noise from the signal that you receive.

 A typical example of this application is the

communication between a pilot, who is inside a jet aircraft,
and a ground control tower. A jet engine can produce noise of
over 140 dB(DECIBELS Breathing ), but normal human
speech is below 50 dB.
Cont...
Definition of Decibels:-
A Decibel is a measurement of Sound
level. Our ears detect changes in volume in a
non-linear fashion.

Definition of Breathing :-
A Breathing in a wider sense, the sound
caused by the friction of the outgoing breath in
the , mouth etc.
 Cont...
 If you are in the ground control tower,
you might have difficulty hearing the pilot's
speech clearly. In this situation, you can use
adaptive filters to remove the jet engine noise
while retaining the pilot's speech.

 The following figure shows a diagram of

the jet engine noise cancellation system.
Figure:-
 Cont...
• In the previous figure, s(n) is the pilot's
speech that you need to acquire. However, you
cannot acquire s(n) directly. You only can
acquire s(n) + v1(n), where v1(n) is the jet engine
noise.

 You cannot acquire v1(n) directly

either. To remove v1(n) from s(n) + v1(n), you
can use an adaptive filter.
Cont...
 You first must use a sensor to acquire only
the jet engine noise v2(n) and send this signal
into the adaptive filter.

 If you compare the previous figure with

the diagram of the adaptive filter, the signal s(n)
+ v1(n) in the previous figure corresponds to d(n)
in the diagram of the adaptive filter. v2(n)
corresponds to x(n).
Cont...
 In the previous figure, if s(n) is
uncorrelated with the jet engine noise v(n), and if
both v1(n) and v2(n) are highly correlated with
v(n), the noise cancellation system can estimate
the jet engine noise v1(n) by adjusting the
coefficients of the adaptive filter iteratively.

 When the output signal y(n) becomes

close to v1(n), the system then can remove the
engine noise. In the previous figure, e(n) denotes
the resulting signal that is close to s(n).
3). Signal Prediction
• It is also known as linear prediction.

• It is a mathematical operation where future

values of a discrete time signal are estimated as
a linear funtion of previous sample.
4). Adaptive feedback
cancellation
• Adaptive feedback cancellation is a application
echo cancellation this is used in telephony to describe
the process of removing echo communication in order
to improve voice quality on a telephone call

• Adaptive feedback cancellation is a common

method of cancelling acoustic feedback in a variety of
electro-acoustic systems such as digital hearing aids.
Difference Traditional Digital
Filter and Adaptive Filter
Traditional Digital Adaptive Filter
Filter 1.It requires additional
1.Has only one input input signal (Desired
signal and single Output)
output signal 2.Error calculation can
2.No error calculation be done
3.Coefficients don’t 3.Coefficients change
change as no error many times as error
calculated. calcuated
•Adaptive filters can complete some signal
processing tasks that traditional digital filters
cannot. For example, you can use adaptive filters
to remove noise that traditional digital filters
cannot remove.
•Adaptive filters can complete some real-time or
online modeling tasks that traditional digital filters
cannot. For example, you can use adaptive filters
to identify an unknown system in online mode.