Introduction

An adaptive filter is a filter that self-adjusts its transfer function according to an optimization algorithm driven by an error signal. Because of the complexity of the optimization algorithms, most adaptive filters are digital filters. By way of contrast, a nonadaptive filter has a static transfer function. Adaptive filters are required for some applications because some parameters of the desired processing operation (for instance, the locations of reflective surfaces in a reverberant space) are not known in advance. The adaptive filter uses feedback in the form of an error signal to refine its transfer function to match the changing parameters. Generally speaking, the adaptive process involves the use of a cost function, which is a criterion for optimum performance of the filter, to feed an algorithm, which determines how to modify filter transfer function to minimize the cost on the next iteration. As the power of digital signal processors has increased, adaptive filters have become much more common and are now routinely used in devices such as mobile phones and other communication devices, camcorders and digital cameras, and medical monitoring equipment.

What is an Adaptive Filter?

An adaptive filter is a computational device that attempts to model the relationship between two signals in real time in an iterative manner. Adaptive filters are often realized either as a set of program instructions running on an arithmetical processing device such as a microprocessor or DSP chip, or as a set of logic operations implemented in a fieldprogrammable gate array (FPGA) or in a semicustom or custom VLSI integrated circuit.
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However, ignoring any errors introduced by numerical precision effects in these implementations, the fundamental operation of an adaptive filter can be characterized independently of the specific physical realization that it takes.

Adaptive filter Aspects:

An adaptive filter is defined by four aspects: 1. The signals being processed by the filter 2. The structure that defines how the output signal of the filter is computed from its input signal 3. The parameters within this structure that can be iteratively changed to alter the filters input-output relationship. 4. The adaptive algorithm that describes how the parameters are adjusted from one time instant to the next.

By choosing a particular adaptive filter structure, one specifies the number and type of parameters that can be adjusted. The adaptive algorithm used to update the parameter values of the system can take on a myriad of forms and is often derived as a form of optimization procedure that minimizes an error criterion that is useful for the task at hand.

In this section, we present the general adaptive filtering problem and introduce the mathematical notation for representing the form and operation of the adaptive filter. We then discuss several different structures that have been proven to be useful in practical applications. We provide an overview of the many and varied applications in which adaptive filters have been successfully used. Finally, we give a simple derivation of the least-mean-square (LMS) algorithm, which is perhaps the most popular method for

adjusting the coefficients of an adaptive filter, and we discuss some of this algorithms properties.

The Adaptive Filtering Problem:

Figure1 shows a block diagram in which a sample from a digital input signal is fed

into a device, called an adaptive filter that computes a corresponding output signal sample at time n. For the moment, the structure of the adaptive filter is not important; except for the fact that it contains adjustable parameters whose values affect how output signal is compared to a second signal is computed. The

, called the desired response signal, by

subtracting the two samples at time n. This difference signal, given by:

isknown as the error signal. The error signal is fed into a procedure which alters or adapts the parameters of the filter from time n to time (n+1) in a well-defined manner. This process of adaptation is represented by the oblique arrow that pierces the adaptive filter block in the figure. As the time index n is incremented, it is hoped that the output of the adaptive filter becomes a better and better match to the desired response signal through this adaptation process, such that the magnitude of decreases over time. In this context,

what is meant by better is specified by the form of the adaptive algorithm used to adjust the parameters of the adaptive filter. In the adaptive filtering task, adaptation refers to the method bywhich the parameters of the system are changed from time index n to time index (n+1). The number and types of parameters within this system depend on the computational structure chosen for the system.
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We now discuss different filter structures that have been proven useful for adaptive filtering tasks.

Adaptive filter problem Figure 1

The Task of an Adaptive Filter:

When considering the adaptive filter problem as illustrated in Fig.1 for the first time, a reader is likely to ask, If we already have the desired response signal. The point of trying to match it using an adaptive filter. In fact, the concept of matching to with

some system obscures the subtlety of the adaptive filtering task. Consider the following issues that pertain to many adaptive

Filtering problems:

In practice, the quantity of interest is not always represent in a certain component of

.Our desire may be to or it may

that is contained in

be to isolate a component of

within the error

that is not contained in

. Alternatively, we may be solely interested in the values of the parameters in and have no concern about , or themselves.

There are situations in which

is not available at all times. In such situations, is available. When is unavailable, in an attemptto

adaptation typically occurs only when

we typically use our most-recent parameter estimates tocompute estimate the desired response signal There are real-world situations in which

is never available. In such cases, one

can use additional information about the characteristics of a hypothetical such as its predicted statistical behavior or amplitude characteristics, to form suitable estimates of from the signals available to the adaptive filter. Such

methods are collectively called blind adaptation algorithms. The fact that such schemes even work is a tribute both to the ingenuity of the developers of the algorithms and to the technological maturity of the adaptive filtering field.

It should also be recognized that the relationship between

and

can vary with

time. In such situations, the adaptive filter attempts to alter its parameter values to follow the changes in this relationship as encoded by the two sequences behavior is commonly referred to as tracking. and this

Applications of Adaptive Filters:

Perhaps the most important driving forces behind the developments in adaptive filters throughout their history have been the wide range of applications in which such systems can be used. We now discuss the forms of these applications in terms of more-general problem classes that describe theassumed relationship between and our

discussion illustrates the key issues in selecting an adaptive filter for a particular task.

System Identification:
Consider Fig. 2, which shows the general problem of system identification. In this diagram, the system enclosed by dashed lines is a black box, meaning that the quantities inside are not observable from the outside. Inside this box is (1) an unknown system which represents a general input output relationship and (2) the signal called the observation noise

signal because it corrupts the observations of the signal at the output of the unknown system.

Fig 2 System Identification Let represent the output of the unknown system with x.n/ as its input. Then, the

desired response signal in this model is:

Here, the task of the adaptive filter is to accurately represent the signal If

at its output.

Then the adaptive filter has accuratelymodeled or identified the portion of the unknown system that is driven by since the model typically chosen for the adaptive filter is a

linear filter, the practical goal of the adaptive filter is to determine the best linear model that describes the input-output relationship of the unknown system. Such a procedure makes the most sense when the unknown system is also alinear model of the same structure as the adaptive filter, as it is possible that

For some set of adaptive filter parameters. For ease of discussion, let the unknown system and the adaptive filter both be FIR filters, such that

Where

/ is an optimum set of filter coefficients for the unknown system at time n. such that

In this problem formulation, the ideal adaptation procedure would adjust

as n!1. In practice, the adaptive filter can only adjust approximates

such that

closely

Overtime.The system identification task is at the heart of numerous adaptive filtering applications

Introduction
Digital signal processing (DSP) has been a major player in the current technical advancements such as noise filtering, system identification, and voice prediction. Standard DSP techniques, however, are not enough to solve these problems quickly and obtain acceptable results. Adaptive filtering techniques must be implemented to promote accurate solutions and a timely convergence to that solution.

Adaptive Filtering System Configurations:

There are four major types of adaptive filtering configurations; adaptive system identification, adaptive noise cancellation, adaptive linear prediction, and adaptive inverse system. All of the above systems are similar in the implementation of the algorithm, but different in system configuration. All four systems have the same general parts; an input x(n), a desired result d(n), an output y(n), an adaptive transfer function w(n), and an error signal e(n) which is the difference between the desired output u(n) and the actual output y(n). In addition to these parts, the system identification and the inverse system configurations have an unknown linear system u(n) that can receive an input and give a linear output to the given input.

Adaptive System Identification Configuration:

The adaptive system identification is primarily responsible for determining a discrete estimation of the transfer function for an unknown digital or analog system. The same input x(n) is applied to both the adaptive filter and the unknown system from which the outputs are compared (see figure 3). The output of the adaptive filter y(n) is subtracted from the output of the unknown system resulting in a desired signal d(n). The resulting difference is

an error signal e(n) used to manipulate the filter coefficients of the adaptive system trending towards an error signal of zero.

Figure 3: Adaptive filter identification configuration After a number of iterations of this process are performed, and if the system is designed correctly, the adaptive filters transfer function will converge to, or near to, the unknown systems transfer function. For this configuration, the error signal does not have to go to zero, although convergence to zero is the ideal situation, to closely approximate the given system. There will, however, be a difference between adaptive filter transfer function and the unknown system transfer function if the error is nonzero and the magnitude of that difference will be directly related to the magnitude of the error signal. Additionally the order of the adaptive system will affect the smallest error that the system can obtain. If there are insufficient coefficients in the adaptive system to model the unknown system, it is said to be under specified. This condition may cause the error to converge to a nonzero constant instead of zero. In contrast, if the adaptive filter is over specified, meaning that there are more coefficients than needed to model the unknown system, the error will converge to zero, but it will increase the time it takes for the filter to converge.

Adaptive Noise Cancellation Configuration:

The second configuration is the adaptive noise cancellation configuration as shown in figure 2. In this configuration the input x(n), a noise source N1(n), is compared with a desired signal d(n), which consists of a signal s(n) corrupted by another noise N0(n). The adaptive filter coefficients adapt to cause the error signal to be a noiseless version of the signal s(n).

Figure 4:Adaptive filter noise cancellation Both of the noise signals for this configuration need to be uncorrelated to the signal s(n). In addition, the noise sources must be correlated to each other in some way, preferably equal, to get the best results. Do to the nature of the error signal; the error signal will never become zero. The error signal should converge to the signal s(n), but not converge to the exact signal. In other words, the difference between the signal s(n) and the error signal e(n) will always be greater than zero. The only option is to minimize the difference between those two signals.

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Adaptive Linear Prediction Configuration:

Adaptive linear prediction is the third type of adaptive configuration (see figure 5). This configuration essentially performs two operations. The first operation, if the output is taken from the error signal e(n), is linear prediction. The adaptive filter coefficients are being trained to predict, from the statistics of the input signal x(n), what the next input signal will be. The second operation, if the output is taken from y(n), is a noise filter similar to the adaptive noise cancellation outlined in the previous section. As in the previous section, neither the linear prediction output nor the noise cancellation output will converge to an error of zero. This is true for the linear prediction output because if the error signal did converge to zero, this would mean that the input signal x(n) is entirely deterministic, in which case we would not need to transmit any information at all.

Figure 5: Linear Prediction Configuration In the case of the noise filtering output, as outlined in the previous section, y(n) will converge to the noiseless version of the input signal.

Adaptive Inverse System Configuration:

The final filter configuration is the adaptive inverse system configuration as shown in figure 4. The goal of the adaptive filter here is to model the inverse of the unknown system u(n). This is particularly useful in adaptive equalization where the goal of the filter is to
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eliminate any spectral changes that are caused by a prior system or transmission line. The way this filter works is as follows. The input x(n) is sent through the unknown filter u(n) and then through the adaptive filter resulting in an output y(n). The input is also sent through a delay to attain d(n). As the error signal is converging to zero, the adaptive filter coefficients w(n) are converging to the inverse of the unknown system u(n).

Figure 6: Adaptive Inverse Configuration For this configuration, as for the system identification configuration, the error can theoretically go to zero. This will only be true; however, if the unknown system consists only of a finite number of poles or the adaptive filter is an IIR filter. If neither of these conditions are true, the system will converge only to a constant due to the limited number of zeroes available in an FIR system.

Performance Measures in Adaptive Systems:

Six performance measures will be discussed in the following sections; convergence rate, minimum mean square error, computational complexity, stability, robustness, and filter length.

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Convergence Rate:

The convergence rate determines the rate at which the filter converges to its resultant state. Usually a faster convergence rate is a desired characteristic of an adaptive system. Convergence rate is not, however, independent of all of the other performance characteristics. There will be a tradeoff, in other performance criteria, for an improved convergence rate and there will be a decreased convergence performance for an increase in other performance. For example, if the convergence rate is increased, the stability characteristics will decrease, making the system more likely to diverge instead of converge to the proper solution. Likewise, a decrease in convergence rate can cause the system to become more stable. This shows that the convergence rate can only be considered in relation to the other performance metrics, not by itself with no regards to the rest of the system.
Minimum Mean Square Error:

The minimum mean square error (MSE) is a metric indicating how well a system can adapt to a given solution. A small minimum MSE is an indication that the adaptive system has accurately modeled, predicted, adapted and/or converged to a solution for the system. A very large MSE usually indicates that the adaptive filter cannot accurately model the given system or the initial state of the adaptive filter is an inadequate starting point to cause the adaptive filter to converge. There are a number of factors which will help to determine the minimum MSE including, but not limited to; quantization noise, order of the adaptive system, measurement noise, and error of the gradient due to the finite step size.

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Computational Complexity:

Computational complexity is particularly important in real time adaptive filter applications. When a real time system is being implemented, there are hardware limitations that may affect the performance of the system. A highly complex algorithm will require much greater hardware resources than a simplistic algorithm.
Stability:

Stability is probably the most important performance measure for the adaptive system. By the nature of the adaptive system, there are very few completely asymptotically stable systems that can be realized. In most cases the systems that are implemented are marginally stable, with the stability determined by the initial conditions, transfer function of the system and the step size of the input.
Robustness:

The robustness of a system is directly related to the stability of a system. Robustness is a measure of how well the system can resist both input and quantization noise.
Filter Length:

The filter length of the adaptive system is inherently tied to many of the other performance measures. The length of the filter specifies how accurately a given system can be modeled by the adaptive filter. In addition, the filter length affects the convergence rate, by increasing or decreasing computation time, it can affect the stability of the system, at certain step sizes, and it affects the minimum MSE. If a thefilter length of the system is increased, the number of computations will increase, decreasing the maximum convergence rate. Conversely, if the filter length is decreased, the number of computations will decrease,

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increasing the maximum convergence rate. For stability, due to an increase in length of the filter for a given system, you may add additional poles or zeroes that may be smaller than those that already exist. In this case the maximum step size, or maximum convergence rate, will have to be decrease to maintain stability. Finally, if the system is under specified, meaning there are not enough pole and/or zeroes to model the system, the mean square error will converge to a nonzero constant. If the system is over specified, meaning it has too many poles and/or zeroes for the system model, it will have the potential to converge to zero, but increased calculations will affect the maximum convergence rate possible.
Filter Algorithms:

A number of filter algorithms will be discussed in this section; the finite impulse response (FIR) least mean squares (LMS) gradient approximation method will be discussed in detail, characteristics of infinite impulse response (IIR) adaptive filters will be briefly discussed, the transform domain adaptive filter (TDAF) and numerous other algorithms will be mentioned for completeness.
Finite Impulse Response (FIR) Algorithms:

Least Mean Squares Gradient Approximation Method Given an adaptive filter with an input x(n), an impulse response w(n) and an output y(n) you will get a mathematical relation for the transfer function of the system

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y(n) = wT(n)x(n) and x(n) = [x(n), x(n-1), x(n-2), ... , x(n-(N-1))] where wT(n) = [w0(n), w1(n), w2(n) ... wN-1(n)] are the time domain coefficients for an Nth order FIR filter. Note in the above equation and throughout a boldface letter represents a vector ant the super script T represents the transpose of a real valued vector or matrix. Using an estimate of the ideal cost function the following equation can be derived. w(n+1) = w(n) - E[e2](n). In the above equation w(n+1) represents the new coefficient values for the next time interval, is a scaling factor, and E[e2](n) is the ideal cost function with respect to the vector w(n). From the above formula one can derive the estimate for the ideal cost function w(n+1) = w(n) - e(n)x(n) where e(n) = d(n) - y(n) and y(n) = xT(n)w(n). In the above equation is sometimes multiplied by 2, but here we will assume it is absorbed by the factor. In summary, in the Least Mean Squares Gradient Approximation Method, often referred to as the Method of Steepest Descent, a guess based on the current filter coefficients is made, and the gradient vector, the derivative of the MSE with respect to the filter coefficients, is
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calculated from the guess. Then a second guess is made at the tap-weight vector by making a change in the present guess in a direction opposite to the gradient vector. This process is repeated until the derivative of the MSE is zero.

Convergence of the LMS Adaptive Filter:

The convergence characteristics of the LMS adaptive filter is related to the autocorrelation of the input process as defined by: Rx = E[x(n)xT(n)] There are two conditions that must be satisfied in order for the system to converge. These conditions include: The autocorrelation matrix, Rx, must be positive definite. 0 < < 1/max., where max is the largest eigenvalue of Rx. In addition, the rate of convergence is related to the eigenvalue spread. This is defined using the condition number of Rx, defined as = max/min, where min is the minimum eigenvalue of Rx. The fastest convergence of this system occurs when = 1, corresponding to white noise. This states that the fastest way to train a LMS adaptive system is to use white noise as the training input. As the noise becomes more and more colored, the speed of the training will decrease.

Transform Domain Adaptive Filter (TDAF)

The TDAF refers to the application of an adaptive filter after a transform of the input system to an orthogonal basis, such as the discrete Fourier Transform (DFT), the discrete cosine transform (DCT), and the Walsh Hadamard transform (WHT). The primary

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motivation for the use of the TDAF is that for colored noise, the TDAF can exhibit faster convergence rates. This will be briefly explained in the next section. The TDAF, in simple terms, is a transform of the system, using one of the above techniques, followed by an application of an adaptive filter algorithm, such as the LMS adaptive filter.
Convergence of the TDAF

It can be shown that the optimum MSE surface is a hypersphere. When white noise is used in training, the error surface will approach a hypersphere. If, however, colored noise is being used as the training input, the shape of the MSE surface will be a hyperellipse. This causes an increase in the convergence time. When the TDAF is applied to the system with a colored noise input, it causes a rotation and a scaling of the axis, causing the points of the MSE surface to intersect the axes at equal distances. This will cause an increase in the convergence time of the adaptive filter.

Quasi-Newton Adaptive Algorithms:

The quasi-Newton adaptive algorithm uses second order statistics to reduce the convergence rate of an adaptive filter, via the Gauss-Newton method. Probably the best known quasi-Newton algorithm is the recursive least squares (RLS) algorithm. It is important to note that even with the increase in convergence rate, the RLS algorithm requires great amounts of processing power, which can make it difficult to implement on real-time systems. There are a number of other quasi-Newton algorithms that have fast convergence rates, and that are also feasible alternatives for real time processing.

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Adaptive Lattice Algorithms

The primary reason for the use of a lattice structure is to reduce the quantization noise introduced by filter coefficients in finite word length systems. As the name suggests, adaptive lattice algorithms are design with the goal of reducing coefficient quantization error, and thus, possibly decreasing the word length of the system with comparable performance

Infinite Impulse Response (IIR) Adaptive Filters:

The primary advantage of IIR filters is that to produce an equivalent frequency response to an FIR filter, they can have a fewer number of coefficients. This in theory should reduce the number of adds, multiplies and shifts to perform a filtering operation. This theory of using IIR filters to reduce the computational burden is the primary motivation for the use of IIR adaptive filters. There are, however, a number of problems that are introduced with the use of IIR adaptive filters. The fundamental concern with IIR adaptive filters is the potential for instability due to poles moving outside the unit circle during the training process. Even if the system is initially stable and the final system is stable, there is still the possibility of the system going unstable during the convergence process. Some suggestion has been made to limit the poles to within the unit circle, however, this method requires that the step sizes be small, which considerably reduces the convergence rate. Due to the interplay between the movement of the poles and zeros, the convergence of IIR systems tends to be slow . The result is that even though IIR filters have fewer coefficients, therefore few calculations per iteration, the number of iterations may increase cause a net loss in processing time to convergence. This, however, is not a problem with all pole filters.
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 In an IIR system, the MSE surface may contain local minimum that can cause a convergence of that system to the local minimum instead of the absolute minimum. More care need to be taken in the initial conditions in IIR adaptive filters than in FIR adaptive filters. IIR filters are more susceptible to coefficient quantization error than FIR, due to the feedback. There have been a number of studies done on the use of IIR adaptive filters, but due to the problems stated above, they are still not widely used in industry today.

MATLAB:
MATLAB (matrix laboratory) is a numerical computing environment and fourth-generation programming language. Developed byMathWorks, MATLAB allows matrix manipulations, plotting of functions and data, implementation of algorithms, creation of user interfaces, and interfacing with programs written in other languages, including C, C++, Java, and Fortran. Although MATLAB is intended primarily for numerical computing, an optional toolbox uses the MuPAD symbolic engine, allowing access to symbolic computing capabilities. An additional package, Simulink, adds graphical multi-domain simulation and Model-Based Design for dynamic and embedded systems. In 2004, MATLAB had around one million users across industry and academia. MATLAB users come from various backgrounds ofengineering, science, and economics. MATLAB is widely used in academic and research institutions as well as industrial enterprises.

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The Scenario:
Figure 1 presents a block diagram of system identification application using adaptive filtering. The objective is to change (adapt) the coefficients of a filter W (which can be a FIRor an IIR one), to match as closely as possible the response of an unknown system H. Theunknown system and the adapting filter process the same input signal x[n] and have asoutputs: d[n] (also referred to as the desired signal) and y[n]. The filter W is adapted using the least mean-square algorithm, which is the most widely used adaptive filtering algorithm. First the error signal, e[n], computed as in eq.1,measures the difference between the output of the adaptive filter and the output of theunknown system. On the basis of this measure, the adaptive filter will change its coefficientsc[n] in an attempt to reduce the error. The coefficient update relation is a function of the errorsignal squared and is given by the following equation:

e[n] = d[n] - y[n] c[n+1] = c[n] + e[n] x[n]

Figure 7: The scenario block diagram

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The convergence time of the LMS algorithm depends on the step size . If is small, then it may take a long convergence time and this may defeat the purpose of using an LMSfilter. However if is too large, the algorithm may never converge. The LMS reference design has the following two main functional blocks: the FIR (or IIR) Filter and the LMS AlgorithmThe FIR filter is implemented serially using a multiplier and an adder with a feedbackas shown in the high level schematic from Figure 2. The FIR result is normalized to minimize saturation.

Figure 8: Using Adaptive filter to identify the Unknown system

The MATLAB Code:

Code Explanation:

1. The user firstly input the number of samples of the input signal. Number of samples is held on the variable N. 2. Step size is also requested from the user. The MU factor will later affect the convergence rhythm of the algorithm.

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3. The input signal is added to noise signal and the output signal of this addition will act as a new input signal. 4. The output signal from 3 above is introduced to an unknown system (the function is already known, just to check to what extent the LMS algorithm is working). The output of the system is the desired signal. 5. Adaptive filter initialization through the LMS. The Adaptive filter inputs are: desired signal (output from the unknown system) + input signal (after adding noise) + the LMS factor (Ha). The output of the adaptive filter is the y (which should be close to desired signal) + the error. 6. Last step is to show the marginal differences between the adapted signal and the actual one. Also comparing the errors through time.

Functions used in code:

Filter: The function "filter" is actually a digital filter. It can be low pass or high pass based on the coefficients. Output of the filter = filter(b, a, input signal). b is the denominator of the filter function. a is the nominator of the filter function. Example: you can use filter to find a running average without using a for loop. This example finds the running average of a 16-element vector, using a window size of 5. data = [1:0.2:4]'; windowSize = 5; filter(ones(1,windowSize)/windowSize,1,data) ans = 0.2000

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0.4400 0.7200 1.0400 1.4000 1.6000 1.8000 2.0000 2.2000 2.4000 2.6000 2.8000 3.0000 3.2000 3.4000 3.6000 Algorithm: The filter function is implemented as a direct form II transposed structure

Where n-1 is the filter order, and which handles both FIR and IIR filters [1]. The operation of filter at sample is given by the time domain difference equations:
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The input-output description of this filtering operation in the -transform domain is a rational transfer function. Randn: Normally distributed random numbers Syntax: Y = randn Y = randn(n) Y = randn(m,n) Y = randn([m n]) Y = randn(m,n,p,...) Y = randn([m n p...]) Y = randn(size(A)) randn(method,s) s = randn(method) Description

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Y = randn returns a pseudorandom, scalar value drawn from a normal distribution with mean 0 and standard deviation 1. Y = randn(n) returns an n-by-n matrix of values derived as described above. Y = randn(m,n) or Y = randn([m n]) returns an m-by-n matrix of the same. Y = randn(m,n,p,...) or Y = randn([m n p...]) generates an m-by-n-by-p-by-... array of the same. Example R = randn(3,4) might produce R= 1.1650 0.3516 0.0591 0.8717

0.6268 -0.6965 0.0751 Fircband: 1.6961

1.7971 -1.4462 0.2641 -0.7012

Perform constrained-band equiripple FIR filter design Syntax: b = fircband(n,f,a,w,c) b = fircband(n,f,a,s) b = fircband(...,'1') b = fircband(...,'minphase')

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b = fircband(..., 'check') b = fircband(...,{lgrid}) [b,err] = fircband(...) [b,err,res] = fircband(...) Description: fircband is a minimax filter design algorithm that you use to design the following types of real FIR filters: Types 1-4 linear phase Type 1 is even order, symmetric Type 2 is odd order, symmetric Type 3 is even order, antisymmetric Type 4 is odd order, antisymmetric Minimum phase Maximum phase, Minimum order (even or odd), extra ripple Maximal ripple Constrained ripple Single-point band (notching and peaking) Forced gain Arbitrary shape frequency response curve filters b = fircband(n,f,a,w,c) designs filters having constrained error magnitudes (ripples). c is a cell array of strings of the same length as w. The entries of c must be either 'c' to indicate that the corresponding element in w is a constraint (the ripple for that band cannot exceed that value) or 'w' indicating that the corresponding entry in w is a weight. There must be at least one unconstrained band--c must contain at least one w entry. For instance, Example 1 below uses a weight of one in the passband, and constrains the stopband ripple not to exceed 0.2 (about 14 dB). A hint about using constrained values: if your constrained filter does not touch the constraints, increase the error weighting you apply to the unconstrained bands. Notice that, when you work with constrained stopbands, you enter the stopband constraint according to the standard conversion formula for power--the resulting filter attenuation or

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constraint equals 20*log(constraint) where constraint is the value you enter in the function. For example, to set 20 dB of attenuation, use a value for the constraint equal to 0.1. This applies to constrained stopbands only. b = fircband(n,f,a,s) is used to design filters with special properties at certain frequency points. s is a cell array of strings and must be the same length as f and a. Entries of s must be one of: 'n'--normal frequency point. 's'--single-point band. The frequency band is given by a single point. You must specify the corresponding gain at this frequency point in a. 'f'-forced frequency point. Forces the gain at the specified frequency band to be the value specified. 'i'--indeterminate frequency point. Use this argument when bands abut one another (no transition region).

MATLAB CODE (M-file): 1. Clear all the MATLAB environment variables and main screen clc;
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clear all; 2. The user firstly input the number of samples of the input signal. Number of samples is held on the variable N. N= input('No of Samples :'); n=(1:N)'; x= input ('enter signal:'); 3. Step size is also requested from the user. The MU factor will later affect the convergence rhythm of the algorithm. Step=input ('please enter the mu (step size) :'); 4.The input signal is added to noise signal and the output signal of this addition will act as a new input signal. v = 0.8*randn(N,1); ar = [1,1/2]; v1 = filter(1,ar,v); s=x+v1; subplot(4,1,1); 5. Plotting the input signal plot(s); title('Received Signal S(t)'); xlabel('Time'); ylabel('Amplitude'); % received signal % Random noise part. % FILTER coefficients. % Noise signal. Applies a 1-D digital filter. % added with noise.. . %No of Samples

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6. Designing the Unknown system [b,err,res] = fircband(30,[0 0.4 0.9 1],[1 1 0 1],[1 0.2],{'w','c'}); unknown system 7. Calculating the desired signal, the output signal from 3 above is introduced to an unknown system (the function is already known, just to check to what extent the LMS algorithm is working). The output of the system is the desired signal. % designs the

d=filter(b,1,s); subplot(4,1,2); plot(d); title('DESIRED SIGNAL d(t)'); xlabel('Time'); ylabel('Amplitude');

% desired signal

8. Adaptive filter initialization through the LMS. The Adaptive filter inputs are: desired signal (output from the unknown system) + input signal (after adding noise) + the LMS factor(Ha). The output of the adaptive filter is the y ( which should be close to desired signal) + the error. ha=adaptfilt.lms(31,step); [y,e]=filter(ha,s,d); subplot(4,1,3); plot(y); %ADAPTIVE ALGORITHM

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title('Adaptive filter output y(t)'); xlabel('Time'); ylabel('Amplitude'); subplot(4,1,4); plot(e); title('Error signal e(t)'); xlabel('Time'); ylabel('Amplitude'); 9. Last step is to show the marginal differences between the adapted signal and the actual one. Also comparing the errors through time. % Displaying the marginal difference between adapted system and actual % system disp ('Ideal System Coefficients'); disp ([b]); disp ('Derived system (adsaptive filter)'); disp ([ha.coefficients]); disp( 'Difference between two systems'); disp ([ha.coefficients-b]); %%%%%%%%%%%%%%%%%%%%Calculation convergance%%%%%%%%%%%%%%%%%%%% of Point of

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E_db=20*log10(abs(e)); a=0; for n= 16 : N if(E_db(n)<=-100 && a==0) a=1; break; end; end; %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Conv_Sample=n figure(2); plot(1:N,[d,y,e]); title('System Identification'); xlabel('Time'); ylabel('Amplitude'); legend('DESIRED','OUTPUT','ERROR'); figure(3);subplot(2,1,1); stem(b);

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title('ACTUAL SYSTEM'); xlabel('Step'); ylabel('h(k)'); subplot(2,1,2); stem([ha.coefficients]); title('IDENTIFIED SYSTEM'); xlabel('Step'); ylabel('ha(k)'); figure(4); subplot(2,1,1); plot((abs(e))); title('Error'); ylabel('e(n)'); xlabel('iteration number'); subplot(2,1,2); plot(20*log10(abs(e))); title('Error(dB)'); ylabel('20 log e(n)'); xlabel('iteration number'); Results and discussion:
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Figure 9: shows the input signal to the unknown system (received signal), the desired signal, the adaptive filter output and the error signal. As it can be clearly seen that the adaptive filter has successfully constructed the system function. Also, the error signal amplitude was high at the beginning of the identification process but with further iterations the error amplitude has been reduced significantly. It can also be easily recognized that the error reduction has occurred in the very beginning, which proves one of the LMS algorithm regarding fast error reduction.

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Figure: 10. Shows the desired, output and error signals. The above diagram plots the three important signals, the desired signal, the actual system output signal and the error signal. The error signal here is the difference between the desired and the actual output signals. As it can be clearly seen that the desired signal and the actual output signal are in complete match (actually the blue desired is behind the green actual output). And this phenomena happens immediately after the errors diminish.

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Figure 11: shows the frequency domain representation for the actual system and the identified system. It can be clearly seen that there is high similarities between the tow graphs.

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Conclusion: The convergence time of the LMS algorithm depends on the step size . If is small, then it may take a long convergence time and this may defeat the purpose of using an LMSfilter. However if is too large, the algorithm may never converge. The LMS reference design has the following two main functional blocks: the FIR (or IIR) Filter and the LMS Algorithmthe FIR filter is implemented serially using a multiplier and an adder with a feedbackas shown in the high level schematic from Figure 11. The FIR result is normalized to minimize saturation. There are two main reasons why the LMS adaptive filter is so popular: 1. It is relatively easy to be implemented in software and hardware due to itscomputational simplicity and efficient use of memory; 2. It performs robustly in the presence of numerical errors caused by finite-precision arithmetic. For the system identification application, the first step was to create a Simulink modelwhich provided the input signals for the hardware design. Concerning the hardware implementation, the LMS adaptive filter core was coded in Verilog-HDL, a dedicated description language, after a previous created block scheme. Simultaneously. All the test filesand the filter core were instantiated into a top level file to be simulated synchronous. Theresults prove that in time, the coefficients adapt and the error converges.

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