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**What is an Adaptive filter?
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y An Adaptive filter is a filter that self-adjusts

its transfer function according to an optimizing algorithms. y 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)

to update the filter coefficients which is to be minimized.Cont« y 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). .

«. 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) y y y . The coefficients for a filter of order p are defined as Wn = [Wn(0). For such structures the impulse response is equal to the filter coefficients.Discussion of Block Diagram y To start the discussion of the block diagram we take the following assumptions: interfering noise v(n) y x(n) = d(n) + v(n) y y The input signal is the sum of a desired signal d(n) and The variable filter has a Finite Impulse Response (FIR) structure.

.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.

y The variable filter updates the filter coefficients at every time instant Where Wn is a correction factor for the filter coefficients..Cont. The adaptive algorithm generates this correction factor based on the input and error signals. .

Recursive least squares (RLS) Algorithm. . 2.Cont« There are two different algorithm to update the co-efficient of filter:--1. Least Mean Square(LMS) Algorithm.

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. which has an unknown response. Since we have the signal both before and after the influence of the room. Least mean squares(LMS) filter y Why LMS ??? In our system. we can use it to remove the interference of the music from the input to the microphone. the music that is being played by the computer is a known output that is being fed into the room. . Once an estimate is available. and picked up by the microphone.1. .

y[n] and find the filter coefficients that relate to producing the least mean squares of the error signal . and the estimated value of the signal. Essentially. Basically: y[n]= h(n)x T(n) . x[n] . and running it through an approximation of the unknown channel. d[n]. This estimated signal is created by taking the original input.LMS Overview y One of the most popular adaptive algorithms used today is the Least Mean Squares (LMS) algorithm. this algorithm attempts to minimize the error that occurs between the detected (or desired signal).

e[n] is the error in this modeling. v[n] is the interference in the room. e(n)=d(n)-y¶(n) . h[n] is the impulse response(weights). x[n] is the input to the filter..Cont.

the system will not converge at all. If the value is too small. then the adaptive filter will not adapt fast enough.Cont. e*(n)x(n) Choosing µ:-The choice of µ is an important one as it greatly affects how the system will perform. we can approximate the next set of weights as follows: h¶(n+1) = h¶(n) + where µ is a constant.. y Using the error. . If it is too large. and the value weights will actually diverge.

y 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. Recursive least squares (RLS) Filter. . The difference is that RLS filters are dependent on the signals themselves. y In the RLS method. the error considered is the total error from the beginning to the current data point. This is contrast to other algorithms that aim to reduce the mean square error. whereas MSE filters are dependent on their statistics.2.

The error signal e(n) and desired signal d(n) are defined in the negative feedback diagram below: .. y The idea behind RLS filters is to minimize a cost function C by appropriately selecting the filter coefficients wn.Cont. updating the filter as new data arrives.

1.x(n-p)]T a(n) = d(n) .x(n-1).Cont. x(n)=[x(n).w(n-1) T x(n) [Prior error] g(n) = P(n)x*(n) [Gain vector] P(n)= -1P(n-1).«.g(n) x T(n) -1P(n-1) w(n) = w(n-1) + a(n)g(n) .2.«. y The RLS algorithm for a p-th order RLS filter can be summarized as y Parameters: p = filter order = forgetting factor = value to initialize P(0) y Initialization: wn=0 P(0)= -1I where I is the (p + 1)-by-(p + 1) identity matrix y Computation: For n=0..

When = 1. This makes the filter more sensitive to recent samples. applying the factor is equivalent to weighting the older error. all previous error is considered of equal weight in the total error.Cont. y Choosing :-As approaches zero. the past errors play a smaller role in the total.. which means more fluctuations in the filter co-efficients. . Since 0 < 1.

While minimizing the error using LMS algorithm.LMS Algorithm V/s RLS Algorithm paths and identifications are the 1. the error considered is the total error from the beginning to the current data point. Note that the signal same whether the filter uses RLS or LMS. . In the RLS method. The difference lies in the adapting portion. 2. it considers only the current error value.

In contrast to the least mean squares algorithm. from which it can be derived. The LMS algorithm requires approximately 20M iterations to converge in mean square. 3. the RLS adaptive algorithm minimizes the total squared error between the desired signal and the output from the unknown system. .. 4.Cont. faster than that of the LMS algorithm by an order of magnitude. The rate of convergence of the RLS algorithm is therefore. the RLS algorithm converges in mean square within less than 2M iterations. where M is the number of tap coefficients contained in the tapped-delayline filter. in general. On the other band.

e. is attained at the expense of a large increase in computational complexity.Cont. .. if we want something best than we use complex function in the same way here RLS use complex computation One interesting input option that applies to RLS algorithms is not present in the LMS processes ² a forgetting factor. however. 5. . The superior performance of the RLS algorithm compared to the LMS algorithm. that determines how the algorithm treats past data input to the algorithm. i. 6.

Applications of Adaptive filter:1) Channel equalization and Channel- identification 2) Noise cancellation 3) Signal prediction 4) Adaptive Feedback Cancellation .

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.1) Channel equalization and identification It¶s like when we compose music. .

2). It is a method used for reducing unwanted sound. Bigger noise cancellation systems are used for ship engines y Example: When DJ plays music he always wear head phones to avoid unwanted sound. Noise cancellation It is also known as active noise channel or active noise reduction. Antinoise is used to reduce noise at the working environment with ear plugs. .

A typical example of this application is the communication between a pilot. A jet engine can produce noise of over 140 dB(DECIBELS Breathing ). but normal human speech is below 50 dB. and a ground control tower. y .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. who is inside a jet aircraft.

. Our ears detect changes in volume in a non-linear fashion.. . mouth etc. y Definition of Decibels:- A Decibel is a measurement of Sound level.Cont. the sound caused by the friction of the outgoing breath in the . y Definition of Breathing :- A Breathing in a wider sense.

Cont. y If you are in the ground control tower. y . The following figure shows a diagram of the jet engine noise cancellation system. 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.

Figure:- .

In the previous figure... You only can acquire s(n) + v1(n). To remove v1(n) from s(n) + v1(n).Cont. y . You cannot acquire v1(n) directly either. where v1(n) is the jet engine noise. you can use an adaptive filter. s(n) is the pilot's speech that you need to acquire. you cannot acquire s(n) directly. However.

Cont.. 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. y . y You first must use a sensor to acquire only the jet engine noise v2(n) and send this signal into the adaptive filter. v2(n) corresponds to x(n)..

When the output signal y(n) becomes close to v1(n). y . In the previous figure.. y In the previous figure. e(n) denotes the resulting signal that is close to s(n)..Cont. the noise cancellation system can estimate the jet engine noise v1(n) by adjusting the coefficients of the adaptive filter iteratively. and if both v1(n) and v2(n) are highly correlated with v(n). the system then can remove the engine noise. if s(n) is uncorrelated with the jet engine noise v(n).

3). It is a mathematical operation where future values of a discrete time signal are estimated as a linear funtion of previous sample. Signal Prediction It is also known as linear prediction. .

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. .

Error calculation can be done 3. Coefficients change many times as error calcuated . additional input signal (Desired Output) 2. Has only one input signal and single output signal 2. No error calculation 3. It requires Filter 1.Difference Traditional Digital Filter and Adaptive Filter y Traditional Digital y Adaptive Filter 1. Coefficients don¶t change as no error calculated.

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. For example. you can use adaptive filters to identify an unknown system in online mode. .Adaptive filters can complete some signal processing tasks that traditional digital filters cannot.

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