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MULTIRATE ADAPTIVE FILTERING

A PROJECT REPORT Submitted by

NEERAJ LAD PRITESH PRAJAPATI KRUPESH SHAH

080110111020 080110111044 080110111048

In fulfillment for the award of the degree of

BACHELOR OF ENGINEERING
in ELECTRONICS AND COMMUNICATION

G. H. Patel College of Engineering And Technology, VallabhVidyanagar Gujarat Technological University, Ahmedabad
DECEMBER 2011

G. H. Patel College of Engineering and Technology Department of Electronics and Communication 2011

CERTIFICATE

Date: This is to certify that the dissertation entitled MULTIRATE ADAPTIVE FILTERING has been carried out by NEERAJ LAD 0801110111020, PRITESH PRAJAPATI 080110111044, KRUPESH SHAH 080110111048 under my guidance in fulfillment of the degree of Bachelor Communication of Engineering of in Electronics and (7th Semester / 8th Semester) Gujarat Technological

University, Ahmedabad during the academic year 2011-12.

Guide:

Mr. Mayank Ardeshana Assistant Professor G. H. Patel College of Engineering and Technology Vallabh Vidyanagar

Head of the Department

ACKNOWLEDGEMENT

We take immense pleasure in thanking Dr. Chintan Modi, Head - EC Department, GCET College for having permitted us to carry out this project work. We take this opportunity to express our regards and sincere thanks to my advisor and guide Assistant Prof. Mayank Ardeshana, without whose support, this project would not have been possible. His constant encouragement and moral support gave us the motivation to carry out the project successfully. I am also indebted to Mr. Darshak Bhatt for his valuable and timely guidance. The discussions with him helped a lot in developing an in-depth understanding of the topics involved. Finally, yet importantly, we would like to express our heartfelt thanks to our beloved parents for their blessings, our friends/classmates for their help and wishes for the successful accomplishment of this part of the project.

ABSTRACT

The project brief that we were assigned comes under the Signal Processing area of Electronics. Specifically it involves Adaptive Filter Design. Basically an adaptive filter is a filter that can mimic the impulse response of other filters or other unknown systems. An everyday example of adaptive filters is in the telephone system. An overview is presented of several frequency-domain adaptive filters that efficiently process discrete-time signals using block and multirate filtering techniques. These algorithms implement a linear convolution that is equivalent to a block time-domain adaptive filter, or they generate a circular convolution that is an approximation. Both approaches exploit the computational advantages of the FFT.

Adaptive Filters are generally implemented using LMS algorithm. MATLAB would be the programming language of choice here to develop the algorithm and indeed all other algorithms. Once the algorithm is tried and tested in MATLAB it will be translated to the C programming language as a precursor for real time implementation. We will use our knowledge of adaptive filters in developing noise cancelling application. So far we learned the basics and MATLAB implementations of filters and we will proceed with different algorithms to be implemented in MATLAB and we will use DSP kit- TI-DSK 6713.

LIST OF FIGURES

Figure No Figure Description


Figure 2.1 Figure 2.2 Figure 2.3 Figure 2.4 Figure 2.5 Figure 2.6 Figure 2.7 Figure 2.8 Figure 2.9 Figure 2.10 Figure 2.11 Figure 3.1 Figure 3.2 Figure 3.3 Figure 3.4 Figure 4.1 Figure 4.2 Figure 4.3 Figure 4.4 Block diagram of a digital signal processing Block diagram of Convolution Relationship between Impulse response and frequency Ideal LowPass filter response MATLAB implementation of LowPass Filter response Ideal HighPass filter response MATLAB implementation of HighPass Filter response Ideal BandPass filter response MATLAB implementation of BandPass Filter response Ideal Bandstop filter Characteristic MATLAB implementation of Bandstop Filter response Basic block diagram of Adaptive Filter Block diagram of System Identification Block diagram of Linear Prediction Block diagram of Noise Cancellation Least Mean Square Filter using Adaptive Algorithm Block diagram of Transversal Filter Signal flow graph representation of LMS Summary of LMS algorithm using Flow Chart

Page No
2 3 6 10 11 11 12 12 13 13 14 18 21 22 23 24 26 29 31

TABLE OF CONTENTS

Acknowledgement Abstract List of Figures Table of Contents Chapter : 1 Introduction 1.1 1.2 Chapter : 2 2.1 2.2 2.3 Chapter : 3 3.1 3.2 3.3 3.4 3.5 Chapter : 4 Motivation Project specification Fundamentals of signals and systems processing The filtering problem Implementation of filters Introduction of adaptive filters Adaptive filters using block diagram
Approaches to the development of linear adaptive filtering algorithms

i ii iii iv 1 1 2 8 10 15 17
19

Theoretical Background

Adaptive filters

How to Choose an Adaptive Filter Application of adaptive filters

20 20

LMS algorithm 4.1 4.2 4.3 4.4 Introduction to LMS algorithm Detail description of the algorithm Least-mean-square adaptation algorithm Summary of the LMS algorithm 24 24 25 28 31 32 32 33

4.5 Summary of the LMS algorithm Future implementation Conclusion References

Chapter 1 Introduction
1.1 Motivation The use of adaptive filtering is there in almost all communication systems. Also the applications are numerous. Moreover it is a new field for us which motivated us to take this project.

1.2 Project Specification Adaptive filters are widely used in many situations where the characteristics of some filter or other are unknown. One application for these filters is quite an important part of today's telephone system that is, compensating for the echo problem. The echo path can be viewed as a filter with certain impulse response, and the task of the adaptive filter is to mimic this impulse response. Adaptive filter implementation is common in the time domain; however for an unknown system with a very long impulse response it becomes more efficient to implement the filter in the frequency domain. The objective of this project is to investigate the design and implementation of adaptive filter algorithms, with emphasis on their use in different problems. It will be necessary to develop simulations of adaptive filter algorithms in the time and frequency domains and compare them from the point of view of performance and implementation complexity (in particular to determine the cross over point where frequency-domain implementation becomes more efficient). It will then be necessary to translate the frequency-domain adaptive filter into the C programming language. A suitable laboratory test bench involving a real circuit should be constructed in order to demonstrate the functionality in near real-time. With the completion of the project we should have a fully operational adaptive filter based in frequency domain. This filter will be applicable where the length of the unknown system's impulse response is long enough for efficient implementation of frequency

domain adaptive filtering. Time domain adaptive filtering will not be replaced entirely as it is far more efficient for systems with short impulse responses.

Chapter 2 Theoretical Background

2.1. Fundamentals of signals and systems processing 2.1.1. Signals and Systems A signal is a description of how one parameter varies with another parameter. A system is any process that produces an output signal in response to an input signal. Signal processing is the operation/s done on signals. The general block diagram of such a process is shown [1]

2.1.2. Linearity

Fig2.1 Block diagram of a digital signal processing system

A system is called linear if it has two mathematical properties: homogeneity and additivity. If you can show that a system has both properties, then you have proven that the system is linear. Likewise, if you can show that a system doesn't have one or both properties, you have proven that it isn't linear. A third property, shift invariance, is not a strict requirement for linearity, but it is a mandatory property for most DSP techniques. When you see the term linear system used in DSP, you should assume it includes shift invariance unless you have reason to believe otherwise. These

three properties form the mathematics of how linear system theory is defined and used. [1]

2.1.3. Convolution Convolution is a formal mathematical operation, just as multiplication, addition, and integration. Addition takes two numbers and produces a third number, while convolution takes two signals and produces a third signal. Convolution is used in the mathematics of many fields, such as probability and statistics. In linear systems, it is used to describe the relationship between three signals of interest: the input signal, the impulse response, and the output signal. An input signal, x[n], enters a linear system with an impulse response, h[n], resulting in an output signal, y[n]. In equation form: x[n] * h[n] = y[n]. Expressed in words, the input signal convolved with the impulse response is equal to the output signal. Just as addition is represented by the plus, +, and multiplication by the cross, , convolution is represented by the star, *.

Fig2.2 Block diagram of Convolution

The equation that represents the convolution is:

The system output y[n] is the convolution of the input signal x[n] which convolves with the system impulse response h[n]. [1]

2.1.4. Correlation Correlation is a mathematical operation that is very similar to convolution. Just as with convolution, correlation uses two signals to produce a third signal. This third signal is called the cross-correlation of the two input signals. If a signal is correlated with itself, the resulting signal is instead called the autocorrelation. Correlation is the optimal technique for detecting a known waveform in random noise. That is, the peak is higher above the noise using correlation than can be produced by any other linear system. Using correlation to detect a known waveform is frequently called matched filtering. [1]

2.1.5. The Discrete Fourier Transform (DFT) Fourier analysis is a family of mathematical techniques, all based on decomposing signals into sinusoids. The Discrete Fourier Transform is the family member used with digitized signals. Before we get started on the DFT, let's look for a moment at the Fourier transform (FT) and explain why we are not talking about it instead. The Fourier transform of a continuous-time signal x(t) may be defined as:

The DFT, on the other hand, replaces the infinite integral with a finite sum:

A signal can be either continuous or discrete, and it can be either periodic or aperiodic. The combination of these two features generates the four categories: 1) Aperiodic-Continuous This includes, for example, decaying exponentials and the Gaussian curve. These signals extend to both positive and negative infinity without repeating in a periodic pattern. The Fourier Transform for this type of signal is simply called the Fourier Transform. [1] 2) Periodic-Continuous Here the examples include: sine waves, square waves, and any waveform that repeats itself in a regular pattern from negative to positive infinity. This version of the Fourier transform is called the Fourier series. [1] 3) Aperiodic-Discrete These signals are only defined at discrete points between positive and negative infinity, and do not repeat themselves in a periodic fashion. This type of Fourier transform is called the Discrete Time Fourier Transform. [1] 4) Periodic-Discrete These are discrete signals that repeat themselves in a periodic fashion from negative to positive infinity. This class of Fourier Transform is sometimes called the Discrete Fourier Series, but is most often called the Discrete Fourier Transform. [1] 2.1.6. Frequency Response of Systems Systems are analyzed in the time domain by using convolution. A similar analysis can be done in the frequency domain. Using the Fourier transform, every input signal can be represented as a group of cosine waves, each with a specified amplitude and phase shift. Likewise, the DFT can be used to represent every output signal in a similar form. This means that any linear system can be completely described by how it changes the amplitude and phase of cosine waves passing through it. This information is called the system's frequency response. Since both the impulse response and the frequency response contain complete information about the system,

there must be a one-to-one correspondence between the two. Given one, you can calculate the other. The relationship between the impulse response and the frequency response is one of the foundations of signal processing: A system's frequency response is the Fourier Transform of its impulse response. [1] Keeping with standard DSP notation, impulse responses use lower case variables, while the corresponding frequency responses are upper case. Since h[ ] is the common symbol for the impulse response, H[ ] is used for the frequency response. Systems are described in the time domain by convolution, that is: x[n] * h[n] = y[n]. In the frequency domain, the input spectrum is multiplied by the frequency response, resulting in the output spectrum. As an equation: X[f ] H[f ] = Y[f ]. That is, convolution in the time domain corresponds to multiplication in the frequency domain. And convolution in the frequency domain corresponds to multiplication in the time domain. The next figure illustrates the relationship between the impulse response and the frequency response:

Fig2.3 Relationship between Impulse response and frequency response

2.1.7. Inverse DFT The inverse DFT (the IDFT) is given by:

In summary, the DFT is proportional to the set of coefficients of projection onto the sinusoidal basis set, and the IDFT is the reconstruction of the original signal as a superposition of its sinusoidal projections. This basic architecture extends to all linear orthogonal transforms, including wavelets, Fourier transforms, Fourier series, the discrete-time Fourier transform (DTFT), and certain short-time Fourier transforms (STFT).

2.1.8. The Fast Fourier Transform (FFT) The Fast Fourier Transform is another method for calculating the DFT. While it produces the same result as the other approaches, it is incredibly more efficient, often reducing the computation time by hundreds. This is the same improvement as flying in a jet aircraft versus walking! The FFT requires a few dozen lines of code, and it is one of the most complicated algorithms in DSP. [1] 2.1.8.1. How the FFT works By making use of periodicities in the sines that are multiplied to do the transforms, the FFT greatly reduces the amount of calculation required. Here's a little overview. Functionally, the FFT decomposes the set of data to be transformed into a series of smaller data sets to be transformed. Then, it decomposes those smaller sets into even smaller sets. At each stage of processing, the results of the previous stage are combined in special way. Finally, it calculates the DFT of each small data set. For example, an FFT of size 32 is broken into 2 FFT's of size 16, which are broken into 4 FFT's of size 8, which are broken into 8 FFT's of size 4, which are broken into 16 FFT's of size 2. Calculating a DFT of size 2 is trivial. Here is a slightly more rigorous explanation: It turns out that it is possible to take the DFT of the first N/2 points and combine them in a special way with the DFT of the second N/2 points to produce a single N-point DFT. Each of these N/2-point DFTs can be calculated using smaller DFTs in the same way. One (radix-2) FFT begins, therefore, by calculating N/2 2-point DFTs. These are combined to form N/4 4-point DFTs. The next stage produces N/8 8-point DFTs, and so on, until a single N-point DFT is produced. [1]

2.1.8.2. Efficiency of the FFT The DFT takes N^2 operations for N points. Since at any stage the computation required to combine smaller DFTs into larger DFTs is proportional to N, and there are log2(N) stages (for radix 2), the total computation is proportional to N * log2(N). Therefore, the ratio between a DFT computation and an FFT computation for the same N is proportional to N / log 2(n). In cases where N is small this ratio is not very significant, but when N becomes large, this ratio gets very large. (Every time you double N, the numerator doubles, but the denominator only increases by 1.) 2.1.8.3. Some Terminology used in FFT The radix is the size of FFT decomposition. For single-radix FFT's, the transform size must be a power of the radix. FFT's can be decomposed using DFT's of even and odd points, which is called a Decimation-In-Time (DIT) FFT, or they can be decomposed using a first-half / second-half approach, which is called a DecimationIn-Frequency (DIF) FFT. Generally, the user does not need to worry which type is being used. 2.1.8.4. Implementation of FFT Except as a learning exercise, you generally will never have to. Many good FFT implementations are available in C, FORTRAN and other languages, and microprocessor manufacturers generally provide free optimized FFT implementations in their processors' assembly code, therefore, it is not so important to understand how the FFT really works, as it is to understand how to use it.[1] 2.2 The filtering problem Filtering is a process of changing signals spectral content. The change is usually the reduction of certain frequencies in the signal while allowing the other frequencies to pass. A fundamental aspect of signal processing is filtering. Filtering involves the manipulation of the spectrum of a signal by passing or blocking certain portions of the spectrum, depending on the frequency of those portions. Filters are designed according to what kind of manipulation of the signal is required for a particular application. Digital filters

are implemented using three fundamental building blocks: an adder, a multiplier, and a delay element. The design process of a digital filter is long and tedious if done by hand. With the aid of computer programs performing filter design algorithms, designing and optimizing filters can be done relatively quickly. We will use MATLAB, a mathematical software package, to design, manipulate, and analyze digital filters. The design options in MATLAB allow the user to either create a code for designing filters that calls built-in functions. The term filter is often used to describe a device in the form of a piece of physical hardware or software that is applied to a set of noisy data in order to extract information about a prescribed quantity of interest. The noise may arise from a variety of sources. For example, the data 'may have been derived by means of noisy sensors or may represent a useful signal component that has been corrupted by transmission through a communication channel. In any event, we may use a filter to perform three basic information-processing tasks: [2] 1. Filtering is an operation that involves the extraction of information about a quantity of interest at time t by using data measured up to and including time t. 2. Smoothing is different from filtering, in that information about the quantity of interest need not be available at time t, and data measured later than time t can be used in obtaining this information. This means that in the case of smoothing there is a delay in producing the result of interest. Since in the smoothing process we are able to use data obtained not only up to time t but also data obtained after time t, we would expect smoothing to be more accurate in some sense than filtering. 3. Prediction is the forecasting side of information processing. The aim here is to derive information about what the quantity of interest will be like at some time t + in the future, for some > 0, by using data measured up to and including time t. We may classify filters into linear and nonlinear. A filter is said to be linear if the filtered, smoothed, or predicted quantity at the output of the device is a linear function of the observations applied to the filter input. Otherwise, the filter is nonlinear.

Filters used for direct filtering can be either Fixed or Adaptive. [3]
1.

Fixed filters - The design of fixed filters requires a priori knowledge of both the signal and the noise, i.e. if we know the signal and noise beforehand; we can design a filter that passes frequencies contained in the signal and rejects the frequency band occupied by the noise. Filter can be divided into several types which are lowpass filter, highpass filter, bandpass filter and bandstop filter.

2.

Adaptive filters - Adaptive filters, on the other hand, have the ability to adjust their impulse response to filter out the correlated signal in the input. They require little or no a priori knowledge of the signal and noise characteristics. (If the signal is narrowband and noise broadband, which is usually the case, or vice versa, no a priori information is needed; otherwise they require a signal (desired response) that is correlated in some sense to the signal to be estimated.) Moreover adaptive filters have the capability of adaptively tracking the signal under non-stationary conditions.

2.3 Implementation of filters 2.3.1Lowpass filter: The ideal lowpass filter with cutoff frequency used. is shown. Also the practical

filter is shown having cutoff frequency 5K Hz, and sampling frequency of 30K Hz is

Fig2.4 Ideal LowPass filter response

X axis Frequency(Hz) Y axis Magnitude(dB) Cut off freq 5 KHz

X axis Frequency(Hz) Y axis Phase(degrees) Cut off freq 5 KHz

Fig2.5 MATLAB implementation of LowPass Filter response

2.3.2 Highpass filter:

The ideal highpass filter with cutoff frequency 30K Hz is used.

is shown. Also the

practical filter is shown having cutoff frequency 5K Hz, and sampling frequency of

Fig2.6 Ideal HighPass filter response

X axis Frequency(Hz) Y axis Magnitude(dB) Cut off freq 5 KHz

X axis Frequency(Hz) Y axis Phase(degrees) Cut off freq 5 KHz

Fig2.7 MATLAB implementation of HighPass Filter response

2.3.3 Bandpass filter:

The ideal bandpass filter with allowable bandwidth cutoff frequency of 1800, and sampling frequency of 8K Hz is used.

is shown.

Also the practical filter is shown having lower cutoff frequency of 1200 and higher

Fig2.8 Ideal BandPass filter response X axis Frequency(Hz) Y axis Phase(degrees) Cut off freq 1200Hz

X axis Frequency(Hz) Y axis Phase(degrees) Cut off freq 1800Hz

Fig2.9 MATLAB implementation of BandPass Filter response

2.3.4 Bandstop filter:

The ideal bandstop filter with frequencies 1800, and sampling frequency of 8K Hz is used.

is shown. Also the practical

filter is shown having lower cutoff frequency of 1200 and higher cutoff frequency of

Fig2.10 Ideal Bandstop filter Characteristic

X axis Frequency(Hz) Y axis Magnitude(dB) Cut off freq 1800Hz

X axis Frequency(Hz) Y axis Phase(degrees) Cut off freq 2800Hz

Fig2.11 MATLAB implementation of Bandstop Filter response

Chapter 3 Adaptive filters


3.1 Introduction In normal filters, a straight forward approach is used where the "estimate and plug" procedure is implemented. This is a two-stage process whereby the filter first "estimates" the statistical parameters of the relevant signals and then "plugs" the results so obtained into a nonrecursive formula for computing the filter parameters. For real-time operation, this procedure has the disadvantage of requiring excessively elaborate and costly hardware. A more efficient method is to use an adaptive filter. By such a device we mean one that is self-designing in that the adaptive filter relies for its operation on a recursive algorithm, which makes it possible for the filter to perform satisfactorily in an environment where complete knowledge of the relevant signal characteristics is not available. The algorithm starts from some predetermined set of initial conditions, representing whatever we know about the environment. Yet, in a stationary environment, we find that after successive iterations of the algorithm it converges to the optimum Wiener solution in some statistical sense. In a nonstationary environment, the

algorithm offers a tracking capability, in that it can track time variations in the statistics of the input data, provided that the variations are sufficiently slow. As a direct consequence of the application of a recursive algorithm whereby the parameters of an adaptive filter are updated from iteration to iteration, the parameters become data dependent. This, therefore, means that an adaptive filter is in reality a nonlinear device, in the sense that it does not obey the principle of superposition. Notwithstanding this property, adaptive filters are commonly classified as linear or nonlinear. An adaptive filter is said to be linear if the estimate of a quantity of interest is computed adaptively (at the output of the filter) as a linear combination of the available set of observations applied to the filter input. Otherwise, the adaptive filter is said to be nonlinear. A wide variety of recursive algorithms has been developed in the literature for the operation of linear adaptive filters. In the final analysis, the choice of one algorithm over another is determined by one or more of the following factors: Rate of convergence: This is defined as the number of iterations required for the algorithm, in response to stationary inputs, to converge "close enough" to the optimum Wiener solution in the mean-square sense. A fast rate of convergence allows the algorithm to adapt rapidly to a stationary environment of unknown statistics. Misadjustment: For an algorithm of interest, this parameter provides a quantitative measure of the amount by which the final value of the mean-squared error, averaged over an ensemble of adaptive filters, deviates from the minimum mean-squared error that is produced by the Wiener filter. Tracking: When an adaptive filtering algorithm operates in a nonstationary environment, the algorithm is required to track statistical variations in the environment. The tracking performance of the algorithm, however, is influenced by two contradictory features: (1) rate of convergence, and (b) steady-state fluctuation due to algorithm noise. Robustness: For an adaptive filter to be robust, small disturbances (i.e., disturbances with small energy) can only result in small estimation errors. The disturbances may arise from a variety of factors, internal or external to the filter. Computational requirements. Here the issues of concern include (a) the number of operations (i.e., multiplications, divisions, and additions/subtractions) required to make one complete iteration of the algorithm, (b) the size of memory location

required to store the data and the program, and (c) the investment required So program the algorithm on a computer. Structure: This refers to the structure of information flow in the algorithm, determining the manner in which it is implemented in hardware form. For example, an algorithm whose structure exhibits high modularity, parallelism, or concurrency is well suited for implementation using very large-scale integration (VLSI).

Numerical properties. When an algorithm is implemented numerically, inaccuracies are produced due to quantization errors. The quantization errors are due to analog-todigital conversion of the input data and digital representation of internal calculations. Ordinarily, it is the latter source of quantization errors that poses a serious design problem. In particular, there are two basic issues of concern: numerical stability and numerical accuracy. Numerical stability is an inherent characteristic of an adaptive filtering algorithm. Numerical accuracy, on the other hand, is determined by the number of bits (i.e., binary digits) used in the numerical representation of data samples and filter coefficients. An adaptive filtering algorithm is said to be numerically robust when it is insensitive to variations in the wordlength used in its digital implementation. [2]

3.2 Adaptive filters using block diagram 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. 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. For this reason, we shall focus on the mathematical forms of adaptive filters as opposed to their specific realizations in software or hardware. Descriptions of adaptive filters as implemented on DSP chips and on a dedicated integrated circuit can be found as follows: 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-meansquare (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. As for the mathematical notation used throughout this section, all quantities are assumed to be real-valued. Scalar and vector quantities shall be indicated by lowercase (e.g., x) and uppercase-bold (e.g., X) letters, respectively. We represent scalar and vector sequences or signals as x(n) and X(n), respectively, where n denotes the discrete time or discrete spatial index, depending on the application [2]

Fig3.1 Basic block diagram of Adaptive Filter

Figure shows a block diagram in which a sample from a digital input signal x(n) is fed into a device, called an adaptive filter, that computes a corresponding output signal sample y(n) 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 y(n) is computed. The output signal is compared to a second signal d(n), called the desired response signal, by subtracting the two samples at time n. This difference signal, given by: e(n) = d(n) y(n) is known 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 e(n) 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 by which 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. [2] 3.3 Approaches to the development of linear adaptive filtering algorithms

There is no unique solution to the linear adaptive filtering problem. We have to understand the capabilities and limitations of various adaptive filtering algorithms and use this understanding in the selection of the appropriate algorithm for the application at hand. Basically, we may identify two distinct approaches for deriving recursive algorithms for the operation of linear adaptive filters, as discussed. [6] 3.3.1 Stochastic Gradient Approach Here we may use a tapped-delay line or transversal filter as the structural basis for implementing the linear adaptive filter. For the case of stationary inputs, the cost function is also referred to as the index of performance, is defined as the meansquared error. This cost function is precisely a second-order function of the tap weights in the transversal filter. The dependence of the mean-squared error on the unknown tap weights may be viewed to be in the form of a multidimensional paraboloid (i.e., punch bowl) with a uniquely defined bottom or minimum point. As mentioned previously, we refer to this paraboloid as the error-performance surface; the tap weights corresponding to the minimum point of the surface define the optimum solution.

3.3.2 Least-squares Estimation The second approach to the development of linear adaptive filtering algorithms is based on the method of least squares. According to this method we minimize a cost function or index of performance that is defined as the sum of weighted error squares, where the error or residual is itself defined as the difference between some desired response and the actual filter output. The method of least squares may be formulated with block estimation or recursive estimation. In block estimation the input data stream is arranged in the form of blocks of equal length (duration), and the filtering of input data proceeds on a block-by-block basis. In recursive estimation, on the other hand, the estimates of interest (e.g., tap weights of a transversal filter) are updated on a sample-by-sample basis. Ordinarily, a recursive estimator requires less storage than a block estimator, which is the reason for its much wider use in practice. This type of algorithm includes: the standard RLS algorithm, Square-root RLS algorithms and the Fast RLS algorithms. Since these algorithms are hard to understand, they are not used in our project. [6]

3.4 How to Choose an Adaptive Filter Given the wide variety of adaptive filters available it is a choice to be made for an application of interest. The filter used should be based on project type and its computational cost, performance, and robustness are the issues which are to be considered while designing adaptive filters. The use of computer simulation provides a good first step in undertaking a detailed investigation of these issues. We begin by using the LMS algorithm as an adaptive filtering tool for the project. The LMS algorithm is relatively simple to implement. Yet it is powerful enough to evaluate the practical benefits that may result from the application of adaptivity to the problem at hand. Moreover, it provides a practical frame of reference for assessing any further improvement that may be attained through the use of more sophisticated adaptive filtering algorithms. Practical applications of adaptive filtering are very diverse, with each application having peculiarities of its own. The solution for one application may not be suitable for another. Nevertheless, to be successful we have to develop a physical understanding of the environment in which the filter has to operate and thereby relate to the realities of the application of interest. 3.5 Applications of adaptive filters 1) System Identification This is used to identify the response of system. The adaptive filters try to determine the best linear model that describes the input-output relationship of the unknown system.

Fig3.2 Block diagram of System Identification

Let d(n) represent the output of the unknown system with x(n) as its input. Then, the desired response signal in this model is d(n)= d(n)+x(n). Here, the task of the adaptive filter is to accurately represent the signal d(n) at its output. If y(n)= d(n) then the adaptive filter has accurately modeled or identified the portion of the unknown system that is driven by x(n). 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 a linear model of the same structure as the adaptive filter, as it is possible that y(n)= d(n) for some set of adaptive filter parameters. The system identification task is at the heart of numerous adaptive filtering applications. We list some of its applications here. Channel Identification In communication systems, useful information is transmitted from one point to another across a medium such as an electrical wire, an optical fiber, or a wireless radio link. Nonidealities of the transmission medium or channel distort the fidelity of the transmitted signals, making the deciphering of the received information difficult. In such cases, an adaptive filter can be used to model the effects of the channel ISI for purposes of deciphering the received information in an optimal manner. In this problem scenario, the transmitter sends to the receiver a sample sequence x(n) that is known to both the transmitter and receiver. The

receiver then attempts to model the received signal d(n)using an adaptive filter whose input is the known transmitted sequence x(n) After a suitable period of adaptation, the parameters of the adaptive filter in W(n) are fixed and then used in a procedure to decode future signals transmitted across the channel. Plant Identification In many control tasks, knowledge of the transfer function of a linear plant is required by the physical controller so that a suitable control signal can be calculated and applied. In such cases, we can characterize the transfer function of the plant by exciting it with a known signal x(n) and then attempting to match the output of the plant d(n) with a linear adaptive filter. After a suitable period of adaptation, the system has been adequately modeled, and the resulting adaptive filter coefficients in W(n) can be used in a control scheme to enable the overall closed-loop system to behave in the desired manner. [2] 2) Linear Prediction In this system, the input signal x(n) is derived from the desired response signal as x(n)= d(n ), Where is an integer value of delay. In effect, the input signal serves as the desired response signal, and for this reason it is always available. In such cases, the linear adaptive filter attempts to predict future values of the input signal using past samples, giving rise to the name linear prediction for this task.

Fig3.3 Block diagram of Linear Prediction

If an estimate of the signal x(n + ) at time n is desired, a copy of the adaptive filter whose input is the current sample x(n) can be employed to compute this quantity. However, linear prediction has a number of uses besides the obvious application of forecasting future events, as described in the following two applications. [2]

3) Noise cancellation A signal of interest is linearly mixed with other extraneous noises which introduce unacceptable errors in the measurements. Noise cancellation is thus used to cancel this interference from a primary signal.

Fig3.4 Block diagram of Noise Cancellation

Adaptive noise cancelling has been used for several applications. One of the first was a medical application that enabled the electroencephalogram (EEG) of the fetal heartbeat of an unborn child to be cleanly extracted from the much-stronger interfering EEG of the maternal heartbeat signal [2] 4) echo cancellation The term echo cancellation is used in telephony to describe the process of removing echo from a voice communication in order to improve voice quality on a telephone call. In addition to improving subjective quality, this process increases the capacity achieved through silence suppression by preventing echo from traveling across a network. Echo cancellation involves first recognizing the originally transmitted signal that re-appears, with some delay, in the transmitted or received signal. Once the echo is recognized, it can be removed by 'subtracting' it from the transmitted or received signal. This technique is generally implemented using a digital signal processor (DSP), but can also be implemented in software. Echo cancellation is done using either echo suppressors or echo cancellers, or in some cases both. [2]

Chapter 4 LMS algorithm


4.1 Introduction to LMS algorithm Least mean squares (LMS) algorithms are a class of adaptive filter used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean squares of the error signal (difference between the desired and the actual signal). It was invented in 1960 by Stanford University professor Bernard Widrow and his first Ph.D. student, Ted Hoff. The LMS algorithm is an important member of the family of stochastic gradient algorithms. The term "stochastic gradient" is intended to distinguish the LMS algorithm from the method of steepest descent that uses a deterministic gradient in a recursive computation of the Wiener filter for stochastic inputs. A significant feature of the LMS algorithm is its simplicity. Moreover, it does not require measurements of the pertinent correlation functions, nor does it require matrix inversion. Indeed, it is the simplicity of the LMS algorithm that has made it the standard against which other adaptive filtering algorithms are benchmarked. [6] 4.2 Problem formulation

Fig4.1 Least Mean Square Filter using Adaptive Algorithm

Most linear adaptive filtering problems can be formulated using the block diagram above. That is, an unknown system is to be identified and the adaptive filter attempts

to adapt the filter

to make it as close as possible to

, while using only observable

signals x(n), d(n) and e(n); but y(n), v(n) and h(n) are not directly observable. Its solution is closely related to the Wiener filter. Definition of symbols

d (n) = y (n) + (n) Idea The idea behind LMS filters is to use steepest descent to find filter weights which minimize a cost function. [6] 4.3 Overview of the least-mean-square algorithm The least-mean-square (LMS) algorithm is a linear adaptive filtering algorithm that consists of two basic processes: 1. A filtering process, which involves (a) computing the output of a transversal filter produced by a set of tap inputs, and (b) generating an estimation error by comparing this output to a desired response. 2. An adaptive process, which involves the automatic adjustment of the tap weights of the filter in accordance with the estimation error. Thus, the combination of these two processes working together constitutes a feedback loop around the LMS algorithm, as illustrated in the block diagram of Figure. First, we have a transversal filter, around which the LMS algorithm is built; this component is responsible for performing the filtering process. Second, we have a mechanism for performing the adaptive control process on the tap weights of the transversal filter, hence the designation "adaptive weight-control mechanism" in Figure (a). Details of the transversal filter component are presented in Figure (b). The tap inputs u(n), u(n - 1) ....., u(n - M + 1) form the elements of the M-by-1 tap-input vector u(n), where M - 1 is the number of delay elements; these tap inputs span a multidimensional space denoted by . Correspondingly, the tap weights

form the elements of the M-by- 1 tap-weight vector W(n). The value computed for the tap-weight vector W(n) using the LMS algorithm represents an

estimate whose expected value approaches the Wiener solution W0 (for a wide-sense stationary environment) as the number of iterations n approaches infinity.

Fig4.2 Block diagram of Transversal Filter

During the filtering process the desired response d(n) is supplied for processing, alongside the tap-input vector u(n). Given this input, the transversal filter produces an output used as an estimate of the desired response d(n). Accordingly, we may define an estimation error e(n) as the difference between the desired response and the actual filter output, as indicated in the output end of Figure (b). The estimation error e(n) and the tap-

input vector u(n) are applied to the control mechanism, and the feedback loop around the tap weights is thereby closed. A scalar version of the inner product of the estimation error e(n) and the tap input u(n - k) is computed for k = 0, 1, 2 ..... M - 2, M - 1. The result so obtained defines the correction Wk(n) applied to the tap weight Wk(n) at iteration n + 1. The scaling factor used in this computation is denoted by , is called the step-size parameter. The LMS algorithm uses the product u(n - k)e*(k) as an estimate of element k in the gradient vector J(n) that characterizes the method of steepest descent. In other words, the expectation operator is missed out from all the paths. Accordingly, the recursive computation of each tap weight in the LMS algorithm suffers from a gradient noise. Earlier we stated that the LMS algorithm involves feedback in its operation, which therefore raises the related issue of stability. In this context, a meaningful criterion is to require that J(n) J( ) as n ,

Where J(n) is the mean-squared error produced by the LMS algorithm at time n, and its final value j( ) is a constant. An algorithm that satisfies this requirement is said to be convergent in the mean square. For the LMS algorithm to satisfy this criterion, the step-size parameter has to satisfy a certain condition related to the Eigen structure of the correlation matrix of the tap inputs. The difference between the final value J( ) and the minimum value the Wiener solution is called the excess mean-squared error attained by

. This difference

represents the price paid for using the adaptive (stochastic) mechanism to control the tap weights in the LMS algorithm in place of a deterministic approach as in the method of steepest descent. The ratio of to is called the misadjustment, which is a measure

of how far the steady-state solution computed by the LMS algorithm is away from the Wiener solution. It is important to realize, however, that the Misadjustment, M is under the designer's control. In particular, the feedback loop acting around the tap weights behaves like a low-pass filter, whose "average" time constant is inversely proportional to the stepsize parameter . Hence, by assigning a small value to the adaptive process is made to progress slowly, and the effects of gradient noise on the tap weights ate largely filtered out. This, in turn, has the effect of reducing the Misadjustment.

Therefore the LMS algorithm is simple in implementation, yet capable of delivering high performance by adapting to its external environment. To do so, however, we have to pay particular attention to the choice of a suitable value for the step-size parameter . [6] 4.4 Least-mean-square adaptation algorithm If it were possible to make exact measurements of the gradient vector J(n) at each

iteration n, and if the step-size parameter is suitably chosen, then the tap-weight vector computed by using the steepest-descent algorithm would indeed converge to the optimum Wiener solution. In reality, however, exact measurements of the gradient vector are not possible since this would require prior knowledge of both the correlation matrix R of the tap inputs and the cross-correlation vector P between the tap inputs and the desired response. Consequently, the gradient vector must be estimated from the available data To develop an estimate of the gradient vector J(n), the most obvious strategy is to substitute estimates of the correlation matrix R and the cross-correlation vector P in the formula:

The simplest choice of estimators for R and p is to use instantaneous estimates that are based on sample values of the tap-input vector and desired response, as defined by, respectively, And Correspondingly, the instantaneous estimate of the gradient vector is J(n) = - 2u(n)d*(n) + 2u(n) (n)W(n)

Generally speaking, this estimate is biased because the tap-weight estimate vector W(n) is a random vector that depends on the tap-input vector u(n). Note that the estimate J(n) may also be viewed as the gradient operator applied to the instantaneous squared error

Substituting the estimate of above equation, for the gradient vector vector: W(n + 1) =W(n) + u(n)[d*(n) (n)W(n)]

J(n) in the

steepest descent algorithm, we get a new recursive relation for updating the tap-weight

Here we have used a hat over the symbol for the tap-weight vector to distinguish it from the value obtained by using the steepest-descent algorithm. Equivalently, we may write the result in the form of three basic relations as follows:

1. Filter output: 2. Estimation error: 3. Tap-weight adaptation: The (1) and (2) equations define the estimation error e(n), the computation of which is based on the current estimate of the tap-weight vector, W(n). Note also that the second term, u(n)e*(n), on the right-hand side of Eq. (3) represents the correction that is applied to the current estimate of the tap-weight vector, W(n). The iterative procedure is started with an initial guess W(0). The algorithm described by Equations (1) to (3) is the complex form of the adaptive least-mean-square (LMS) algorithm. At each iteration or time update, it also requires knowledge of the most recent values: u(n), d(n), and W(n). The LMS algorithm is a member of the family of stochastic gradient algorithms. In particular, when the LMS algorithm operates on stochastic inputs, the allowed set of directions along which we "step" from one iteration cycle to the next is quite random and cannot therefore be thought of as being true gradient directions.

Fig4.3 Signal flow graph representation of LMS algorithm

The above figure shows a signal-flow graph representation of the LMS algorithm in the form of a feedback model. This model bears a close resemblance to the feedback model of the steepest-descent algorithm. The signal-flow graph of above figure clearly illustrates the simplicity of the LMS algorithm. In particular, we see from this figure that the LMS algorithm requires only 2M + 1 complex multiplications and 2M complex additions per iteration, where M is the number of tap weights used in the adaptive transversal filter. In other words, the computational complexity of the LMS algorithm is O(M). The instantaneous estimates of R and P have relatively large variances. At first sight, it may therefore seem that the LMS algorithm is incapable of good performance since the algorithm uses these instantaneous estimates. However, we must remember that the LMS algorithm is recursive in nature, with the result that the algorithm itself effectively averages these estimates, in some sense, during the course of adaptation. [6]

4.5 Summary of the LMS algorithm Parameters: M = number of taps = step-size parameter

Tap-input power = Flowchart:

Fig4.4 Summary of LMS algorithm using Flow Chart

Here, And

FUTURE IMPLEMENTATION

We are going to implement the adaptive filters in MATLAB, which can process on sound and then we will go for image. Then we are going to implement it on hardware (DSP kit TI-DSK 6713). We will try to implement this adaptive filter on some real time application.

CONCLUSION
Through this project we learnt various concepts of Digital Signal Processing. We also learnt in detail about adaptive filters and its application. Adaptive filters are those who adapt themselves according to required situation. Adaptive filters use LMS algorithm for error detection. LMS algorithm is simple to understand and easy to implement. Thus we have satisfactorily completed our first part of project.

REFERENCES
Books: 1. Proakis, John G. and Manolakis, Dimitris G. Digital Signal Processing: Principles, Algorithms, and Applications, 3rd Edition. Prentice Hall. Upper Saddle River, NJ,1996 2. Vijay K. Madisetti and Douglas B. Williams Boca Raton (1999) Introduction to Adaptive Filters Digital Signal Processing Handbook 3. Oppeheim, Schafer, Buck Pearson education publication, 2nd Edition (2003) Discrete Time Signal Processing 4. 5. IEEE Transactions on Signal Processing, Vol. 39,No. 10, (1991) E.Ifeachor and B.Jervis, 2nd edition Prentice Hall (1998) Digital Signal Processing: A practical Approach 6. Simon Hayking, 3rd Edition Filter Theory

Papers: 1. John J. Shynk, IEEE SP Magazine Frequency Domain and multirate Adaptive Filtering 2. Christian Feldbauer, Franz Pernkopf, and Erhard Rank , Signal Processing and Speech Communication Laboratory, Inffeldgasse 16c Adaptive Filters Tutorial