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CHAPTER I

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.

areas such as communications, control, and biomedical engineering. Examples of such

applications are adaptive noise canceling, adaptive equalization of data transmission

channels, adaptive antenna arrays, adaptive system identification, and adaptive canceling

of narrowband interference in direct sequence spread spectrum systems.

adaptation algorithm so as to minimize the difference between the filter output and a

reference signal according to some criterion. The nature of the reference signal depends

on the considered application. There are two main measures for evaluating the

performance of an adaptive filter: the convergence rate and the steady state mean square

error. In practical applications, it is desired to maximize the convergence rate and

minimize the steady state mean square error. There is a conflict between these

requirements. Several adaptation algorithms have been developed so as to yield a good

compromise between these requirements. Improving this compromise is a continuous

open issue. This presentation gives a survey of main applications of adaptive filtering,

performance criteria, adaptation algorithms, some new results attained by the presenter,

and open issues.

Adaptive filters sense the properties of the environments in which they operate and

adjust the filter parameters accordingly. Consequently, they are useful in a wide variety

of applications in which the properties of the operating environments are not known, or in

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which they change with time in a previously unknown manner. Adaptive filters, which

aim to transform information-bearing signals into “cleaned up” or “improved” versions,

adjust their characteristics according to the signals encountered. They form the simplest

examples of algorithms within the field of machine learning.

Adaptive filters are often preferred over their fixed-characteristic counterparts, which

are fundamentally unable to adjust to changing signal conditions. The convenient

autonomous adaptability of adaptive filters explains their widespread application in signal

restoration, interference cancellation, system identification, and medical diagnostics, to

name just a few areas. Ubiquitous among the different adaptive filtering algorithm is the

Least-mean-square (LMS) .

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.

synthesis and analysis of adaptive filters (controllers) in “Filtered” LMS problems. This

new approach uses an estimation interpretation of the adaptive filtering (control) problem

to formulate an equivalent estimation problem. The new algorithm, referred to as

Estimation-Based Adaptive Filtering (EBAF),applies to both Finite Impulse Response

(FIR) and Infinite Impulse Response (IIR)adaptive filters.

1.1 MOTIVATION:

Introducing a systematic procedure for the synthesis of adaptive filters is one of the

main goals of this project.Least-Mean Squares (LMS) adaptive algorithm has been the

centerpiece of a wide variety of adaptive filtering techniques for almost four decades. The

straightforward derivation and the simplicity of its implementation (especially at the time

of limited computational power) encouraged experiments with the algorithm in a diverse

range of applications. In some applications however, the simple implementation of the

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shortcomings have produced a large number of innovative solutions that have been

successful in practice.

Commonly used algorithms such as normalized LMS, correlation LMS, leaky LMS,

variable-step-size LMS, and Filtered-X LMS are the outcome of such efforts. These

algorithms use the instantaneous squared error to estimate the mean-square error, and

often assume slow adaptation to allow for the necessary linear operations in their

derivation.

“Many of the algorithms and approaches used are of an ad hoc nature; the tools are

gathered from a wide range of fields; and good systematic approaches are still lacking.”

the adaptation with LMS on stationary stochastic processes, and finds the optimal solution

to which the expected value of the weight vector converges. For sinusoidal inputs, when

time-varying component of the adaptive filter output is small compared to its time-

invariant component, the adaptive LMS filter can be approximated by a linear time-

invariant transfer function.

To put this thesis in perspective, this section provides a brief overview of the vast

literature on adaptive filtering (control). Reference recognizes 1957 as the year for the

formal introduction of the term “adaptive system” into the control literature. By then, the

interest in filtering and control theory had shifted towards increasingly more complex

systems with poorly characterized (possibly time varying) models for system dynamics

and disturbances, and the concept of “adaptation” (borrowed from living systems) seemed

to carry the potential for solving the increasingly more complex control problems.

The exact definition of “adaptation” and its distinction from “feedback”, however, is the

subject of long standing discussions .Qualitatively speaking, an adaptive system is a

system that can modify its behavior in response to changes in the dynamics of the system

or disturbances through some recursive algorithm. As a direct consequence of this

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recursive algorithm (in which the parameters of the adaptive system are adjusted using

input/output data), an adaptive system is a “nonlinear” device. The development of

adaptive algorithms has been pursued from a variety of view Points.

For the purpose of this thesis is to provide a distinct approach for deriving recursive

adaptive algorithm can be identified as stochastic gradient approaches that include LMS

and LMS-Based adaptive algorithm. The central idea in the this approach, is to define an

appropriate cost function that captures the success of the adaptation process, and then

change the adaptive filter parameters to reduce the cost function according to the method

of steepest descent. This requires the use of a gradient vector (hence the name), which in

practice is approximated using instantaneous data.. The latter approach to the design of

adaptive filters is based on the method of least squares. This approach closely corresponds

to Kalman filtering, provides a unifying state-space approach to adaptive RLS filtering.

The main focus in this thesis however, is on the LMS-based adaptive algorithms.

they have been used in a diverse field of applications that include communication, process

control ,seismology, biomedical engineering. Despite the diversity of the applications, die

rent implementations of adaptive filtering (control) share one basic common feature: “an

input vector and a desired response are used to compute an estimation error, which is in

turn used to control the values of a set of adjustable filter coefficients.”

There are several main structures for the implementation of adaptive filters

(controllers). The structure of the adaptive filter is known to affect its performance,

computational complexity, and convergence. In this thesis, the two most commonly used

structures for adaptive filters (controllers) are considered. The finite impulse response

(FIR) transversal filter is the structure upon which the main presentation of the

estimation-based adaptive filtering algorithm is primarily presented. The transversal filter

consists of three basic elements: (a) unit-delay element,

(b) Multiplier and

(c) adder

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and contains feed forwards paths only. The numbers of unit-delays specify the length of

the adaptive FIR filter. Multipliers weight the delayed versions of some reference signal,

which are then added in the adder(s). The frequency response for this filter is of finite

length and contains only zeros (all poles are at the origin in the z-plane). Therefore, there

is no question of stability for the open-loop behavior of the FIR filter. The feature that

distinguishes the IIR filter from an FIR filter is the inclusion of the feedback path in the

structure of the adaptive filter.

As mentioned earlier, for an FIR filter all poles are at the origin, and a good

approximation of the behavior of a pole, in general, can only be achieved if the length of

the FIR filter is sufficiently long. An IIR filter, ideally at least, can provide a perfect match

for a pole with only a limited number of parameters. This means that for a desired

dynamic behavior the number of parameters in an adaptive IIR filter can be far fewer than

that in its FIR counterpart.

The computational complexity per sample for adaptive IIR filter design can therefore

be significantly lower than that in FIR filter design. The limited use of adaptive IIR filters

suggests that the above mentioned advantages come at a certain cost. In particular,

adaptive IIR filters are only conditionally stable, and therefore some provisions are

required to assure stability of the filter at each iteration.

communication channels, and active control of sound and vibration in an environment

where the effect of a number of primary sources should be canceled by a number of

control (secondary) sources, the use of a multi-channel adaptive algorithm is well

justified. In general, however, variations of the LMS algorithm are not easy to extend to

multi-channel systems. Furthermore, the analysis of the performance and properties of

such multi-channel algorithms is complicated . In the context of active noise cancellation,

the successful implementation of multi-channel adaptive algorithms has so far been

limited to cases involving repetitive noise with a few harmonics .

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CHAPTER II

2.1 ADAPTIVE FILTERING:

Adaptive filters are digital filters capable of self adjustment. These filters can change

in accordance to their input signals. An adaptive filter is used in applications that require

differing filter characteristics in response to variable signal conditions.

The adaptive filter requires two inputs:

_ the input signal x(n)

_ the reference input d(n)

An adaptive filter has the ability to update its coefficients. New coefficients are sent to

the filter from a coefficient generator. The coefficient generator is an adaptive algorithm

that modifies the coefficients in response to an incoming signal. The digital filter is

typically a special type of finite impulse response (FIR) filter, but it can be also an

infinite impulse response (IIR) or other type of filter. Adaptive filters have uses in a

number of applications, including noise cancellation, linear prediction, adaptive signal

enhancement, and adaptive control.

Figure 1 shows a block diagram of an adaptive filter model. The unknown system is

modeled by an FIR filter with adjustable coefficients. Both the unknown system and FIR

filter model are excited by an input sequence x(n). The adaptive FIR filter output y(n) is

compared with the unknown system output d(n) to produce the error signal e(n). The

error signal represents the difference between the unknown system output and the model

output. The error e(n) is then used as the input to an adaptive control algorithm, which

corrects the individual tap weights of the filter. This process is repeated through several

iterations until the error signal e(n) becomes sufficiently small. The resultant FIR filter

response now represents that of the previously unknown system.

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Figure 2 shows the structure of a transversal FIR filter with N taps adjustable weights.

w(n) = [w0(n), w0(n), …, w0(n)]

the tap-input vector, x(n), as

x(n) = [x(n), x(n-1), …, x(n-N+1)]

The FIR filter output, y(n) can then be expressed as

N-1

Y(n)= ∑ wi(n). x(n-i).

n=0

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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.

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

y(n) = wT(n)x(n)

and

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) - µ ∆ 2

(n).

E[e ]

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In the above equation w(n+1) represents the new coefficient values for the next time

interval, µ is a scaling factor, and ∆ 2

(n) is the ideal cost function with respect to the

E[e ]

vector w(n). From the above formula one can derive the estimate for the ideal cost

function

where

and

y(n) = xT(n)w(n).

absorbed by the µ factor.

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 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.

autocorrelation of the input process as defined by

Rx = E[x(n)xT(n)]

There are a two conditions that must be satisfied in order for the system to converge.

These conditions include:

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o 0 < µ < 1/λ max., where λ max is the largest eigen value of Rx.

In addition, the rate of convergence is related to the eigen value spread. This is defined

using the condition number of Rx, defined as κ = λ max /λ min , where λ min is the

minimum eigen value of Rx. The fastest convergence of this system occurs when κ = 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.

The LMS algorithm is initialised by setting all weights to zero at time n=0. Tap

weights are updated using the relationship

where

wi(n) represent the tap weights of the FIR filter

e(n) is the error signal

x(n) represent the tap inputs

the factor µ= is the adaptation parameter or step-size

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Adaptive filtering techniques are used in a wide range of applications, including echo

Cancellation, adaptive equalization, adaptive noise cancellation, and adaptive beam

forming. The performance of an adaptive filtering algorithm is evaluated based on its

convergence rate, mis adjustment, computational requirements, and numerical robustness.

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using “unconventional” structures for adaptive filters.

Part I of this dissertation presents a new adaptation algorithm, which we have termed

the Normalized LMS algorithm with Orthogonal Correction Factors (NLMS-OCF). The

NLMSOCF algorithm updates the adaptive filter coefficients (weights) on the basis of

multiple input signal vectors, while NLMS updates the weights on the basis of a single

input vector. The well-known Affine Projection Algorithm (APA) is a special case of our

NLMS-OCF algorithm.

for the input vector. Our analysis shows that the convergence rate of NLMS-OCF (and

also APA) is exponential and that it improves with an increase in the number of input

signal vectors used for adaptation. While we show that, in theory, the misadjustment of

the APA class is independent of the number of vectors used for adaptation, simulation

results show a weak dependence. For white input the mean squared error drops by 20 dB

in about 5N/(M+1) iterations, where N is the number of taps in the adaptive filter and

(M+1) is the number of vectors used for adaptation.

The dependence of the steady-state error and of the tracking properties on the three

user selectable parameters, namely step size µ , number of vectors used for adaptation

(1+ M), and input vector delay D used for adaptation, is discussed. While the lag error

depends on all of the above parameters, the fluctuation error depends only on µ .

Increasing D results in a linear increase in the lag error and hence the total steady state

mean-squared error. The optimum choices for µ and M are derived. Simulation results

are provided to corroborate our analytical results.

We also derive a fast version of our NLMS-OCF algorithm that has a complexity of

O(NM) .The fast version of the algorithm performs orthogonalization using a forward-

backward prediction lattice. We demonstrate the advantages of using NLMS-OCF in a

practical application, namely stereophonic acoustic echo cancellation. We find that

NLMS-OCF can provide faster convergence, as well as better echo rejection, than the

widely used APA.

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While the first part of this project attempts to improve adaptive filter performance by

refining the adaptation algorithm, the second part of this work looks at improving the

convergence rate by using different structures. From an abstract viewpoint, the

parameterization we decide to use has no special significance, other than serving as a

vehicle to arrive at a good input-output description of the system. However, from a

practical viewpoint, the parameterization decides how easy it is to numerically minimize

the cost function that the adaptive filter is attempting to minimize.

condition number for Grammians. Furthermore, a balanced realization is useful in model

order reduction. These properties of the balanced realization make it an attractive

candidate as a structure for adaptive filtering. We propose an adaptive filtering algorithm

based on balanced realizations.

adaptive IIR filtering algorithm. Minimizing the equation error subject to the unit-norm

constraint yields an unbiased estimate for the parameters of a system, if the measurement

noise is white. The proposed algorithm uses the hyper-spherical transformation to convert

this constrained optimization problem into an unconstrained optimization problem. It is

shown that the hyper-spherical transformation does not introduce any new minima in the

equation error surface. Hence, simple gradient-based algorithms converge to the global

minimum. Simulation results indicate that the proposed algorithm provides an unbiased

estimate of the system parameters.

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The Kalman filter is just one of many adaptive filtering (or estimation) algorithms.

Despite its elegant derivation and often excellent performance, the Kalman filter has two

drawbacks:

1. The derivation and hence performance of the Kalman filter depends on the

accuracy of the a priori assumptions. The performance can be less than impressive

if the assumptions are erroneous.

2. The Kalman filter is fairly computationally demanding, requiring O(P2)

operations per sample. This can limit the utility of Kalman filters in high rate real

time applications.

As a popular alternative to the Kalman filter, we will investigate the so-called least-

mean-square (LMS) adaptive filtering algorithm.

The principle advantages of LMS are

2. Computationally, LMS is very efficient, requiring O(P) per sample.

The price we pay with LMS instead of a Kalman filter is that the rate of convergence

and adaptation to sudden changes is slower for LMS than for the Kalman filter (with

correct prior assumptions).

First, simulate the system identification block diagram in Figure 1. For the simulation,

let d [n] = x [n] be a Gaussian random noise input, which can be generated in MATLAB

using the command randn. Because the adaptive filter coefficients change with time,

implement the filter on a sample by sample basis ( e.g., using a do loop). For the

"unknown" system, Use the fourth-order, low-pass elliptical IIR filter designed with the

following MATLAB commands:

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Simulate the system with an adaptive filter of length 32 and a step-size of 0.02.

Initialize all adaptive filter coefficients to zero. Plot the squared-error versus sample

number as the filter adapts over time. Also, plot the frequency response of the adaptive

filter coefficients at the end of the simulation. How well does the filter match? How long

does it take to converge?

Second, write an assembly program to implement an adaptive FIR filter. Your code

can be modified from the FIR filtering laboratory exercise to meet the following

specifications:

1. The program should run at an 8kHz-sampling rate.

2. The filter coefficients should be initialized to zero at the start of program execution.

3. The left input channel is the input x [n] to the adaptive filter.

4. The right input channel is the reference input signal d [n].

5. The left channel output is the adaptive filter output y [n].

6. The right channel output is the error signal e [n].

Based on these specifications, your program should implement the block diagram of

Figure 1.

The LMS filter coefficient update equation is

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where W [n] is the vector of FIR filter coefficients at time n, e [n] = d [n] − y [n] is the

error between the reference and output, µ is the step size, and X [n] is the vector of

current and past input samples.

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FILTERS:

areas such as communications, control, and biomedical engineering. Examples of such

applications are adaptive noise canceling, adaptive equalization of data transmission

channels, adaptive antenna arrays, adaptive system identification, and adaptive canceling

of narrowband interference in direct sequence spread spectrum systems. Adaptive

filtering is made of a digital filter whose weights are controlled by an adaptation

algorithm so as to minimize the difference between the filter output and a reference signal

according to some criterion. The nature of the reference signal depends on the considered

application. There are two main measures for evaluating the performance of an adaptive

filter: the convergence rate and the steady state mean square error. In practical

applications, it is desired to maximize the convergence rate and minimize the steady state

mean square error. There is a conflict between these requirements. Several adaptation

algorithms have been developed so as to yield a good compromise between these

requirements. Improving this compromise is a continuous open issue.

Adaptive filters are ubiquitous tools for numerous real-world scientific and industrial

applications. Many educators and practitioners employ the Matlab/spl reg/ technical

computing environment to implement and study adaptive filters. This paper describes the

design and implementation issues regarding a recently-developed set of comprehensive

Matlab adaptive FIR filtering tools. In addition to a complete suite of algorithms, the tool

set includes analysis functions that enable users to quickly characterize the average

performance of selected algorithms when limited data are available. We provide

execution speed comparisons for algorithm families to guide users in algorithm selection

when Matlab execution time is most critical.

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Adaptive filters sense the properties of the environments in which they operate and

adjust the filter parameters accordingly. Consequently, they are useful in a wide variety

of applications in which the properties of the operating environments are not known, or in

which they change with time in a previously unknown manner.

We have considerable experience in developing and analyzing various types of

adaptive linear and nonlinear filters. Examples of works he has done in this area include:

few or no multipliers for adapting the parameters

(ii) Algorithms that exhibit fast convergence as well as low steady-state

error characteristics

(iii) Efficient techniques for adaptive filters employing nonlinear system

models.

The application areas we are interested in include, but are not limited to:

communication and teleconferencing, compensation of nonlinearities in magnetic

recording media, image binarization for character recognition systems, etc.

• Adaptive prediction in data compression systems including compression of

images, speech, multispectral data, etc.

• Time delay estimation, tracking and localization of moving objects.

• Adaptive enhancement of images.

• Separation of signals generated by several sources from a mixture.

• Adaptive interference cancellation and enhancement of noisy signals.

optimizing algorithm. Because of the complexity of the optimizing algorithms, most

adaptive filters are digital filters that perform digital signal processing and adapt their

performance based on the input signal. By way of contrast, a non-adaptive filter has

static filter coefficients (which collectively form the transfer function).

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For some applications, adaptive coefficients are required since some parameters of the

desired processing operation (for instance, the properties of some noise signal) are not

known in advance. In these situations it is common to employ an adaptive filter, which

uses feedback to refine the values of the filter coefficients and hence its frequency

response.

Generally speaking, the adapting process involves the use of a cost function, which is

a criterion for optimum performance of the filter (for example, minimizing the noise

component of the input), to feed an algorithm, which determines how to modify of the

filter coefficients 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.

Example:

Suppose a hospital is recording a heart beat (an ECG), which is being corrupted by a

50 Hz noise (the frequency coming from the power supply in many countries).

One way to remove the noise is to filter the signal with a notch filter at 50 Hz.

However, due to slight variations in the power supply to the hospital, the exact frequency

of the power supply might (hypothetically) wander between 47 Hz and 53 Hz. A static

filter would need to remove all the frequencies between 47 and 53 Hz, which could

excessively degrade the quality of the ECG since the heart beat would also likely have

frequency components in the rejected range.

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To circumvent this potential loss of information, an adaptive filter could be used. The

adaptive filter would take input both from the patient and from the power supply

directly and would thus be able to track the actual frequency of the noise as it

fluctuates. Such an adaptive technique generally allows for a filter with a smaller

rejection range, which means, in our case, that the quality of the output signal is

more accurate for medical diagnoses.

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 4 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.

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Above figure depicts the system modeling problem. The output of an unknown

system to a known input is used as the reference input to an adaptive filter. The known

input signal is fed through the adaptive filter in an effort to duplicate the unknown system

output. Upon convergence, the adaptive filter parameters represent an estimate of the

unknown system.

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 1). 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.

After a number of iterations of this process are performed, and if the system is

designed correctly, the adaptive filter’s transfer function will converge to, or near to, the

unknown system’s 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.

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Adaptive filters are used in modern telephony to perform echo suppression. A small

amount of speech from talker A, transmitted to talker B, is unavoidably "echoed" back to

talker A with talker B's speech.

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).

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

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greater than zero. The only option is to minimize the difference between those two

signals.

With this application in mind, a stereo speech recording has been made. On one

channel talker A is speaking. On the other channel, speech from talker B is present, along

with a delayed "echo" of talker A's speech. Use your adaptive filter to suppress the

"echo" from the speech of talker B.

Adaptive linear prediction is the third type of adaptive configuration (see figure 3).

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.

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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.

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 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).

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.

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2.6 EQUALIZATION:

In equalization, the received spectral coefficient blocks (i.e. after cyclic prefix

removal and FFT) are adjusted to compensate for the frequency response of the channel

(nothing can be done here about the additive noise). Due to the cyclic prefix1, each block

has essentially undergone cyclic convolution with the channel's impulse response. In the

frequency domain, this is the same as if the spectral coefficients were point wise

multiplied by the frequency response of the channel. If the freq. response has no zeros

and is known by the receiver, it is possible to perfectly remove the effect of the channel's

filter. Since the channel point wise multiplied the blocks by its freq. response, all that

needs to be done is multiply the blocks point wise by the 1 over the freq. response.

Because we implemented the channel's impulse response as non-ideal low-pass, it's freq.

response has no zeros and equalization is rather trivial.

In audio processing, equalization (or equalization, EQ) is the process of changing the

frequency envelope of a sound. In passing through any channel, temporal

/frequency spreading of a signal occurs. Etymologically, it means to correct, or

make equal, the frequency response of a signal. The term "equalizer" is often

incorrectly applied as a general term for audio filters. DJ mixing equipment and

hi-fi audio components often include so called graphic equalizers or simply

equalizer. These are in fact general all-purpose filters, which can be arranged to

produce the effect of low pass, high pass, band pass and band stop filters. Only

when these filters are arranged so as to reverse the effects of Overview.

There are many kinds of EQ. Each has a different pattern of attenuation or boost. A

peeking equalizer raises or lowers a range of frequencies around a central point in a bell

shape. A peaking equalizer with controls to adjust the level (Gain), bandwidth (Q) and

center frequency (Hz) is called a parametric equalizer. If there is no control for the

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parametric equalizer.

A pass filter attenuates either high or low frequencies while allowing other

frequencies to pass unfiltered. A high-pass filter modifies a signal only by taking out low

frequencies; a low-pass filter only modifies the audio signal by taking out high

frequencies. A pass filter is described by its cut-off point and slope. The cut-off point is

the frequency where high or low-frequencies will be removed. The slope, given in

decibels per octave, describes a ratio of how the filter attenuates frequencies past the cut-

off point (eg. 12 dB per octave). A band-pass filter is simply a combination of one high-

pass filter and one low-pass filter which together allow only a band of frequencies to

pass, attenuating both high and low frequencies past certain cut-off points.

by a fixed amount. A low shelf will affect low frequencies up to a certain point and then

above that point will have little effect. A high shelf affects the level of high frequencies,

while below a certain point, the low frequencies are unaffected.

One common type of equalizer is the graphic equalizer, which consists of a bank of

sliders for boosting and cutting different bands (or frequencies ranges) of sound.

Normally, these bands are tight enough to give at least 3 dB or 6 dB maximum effect for

neighboring bands, and cover the range from around 20 Hz to 20 kHz (which is

approximately the range of human hearing). A simple equalizer might have bands at 20

Hz, 200 Hz, 2 kHz and 20 kHz, and might be referred to as a 4-band equalizer. A typical

equalizer for live sound reinforcement might have as many as 24 or 31 bands. A typical

31-band equalizer is also called a 1/3-octave equalizer because the center frequencies of

sliders are spaced one third of an octave apart.

One of the most direct uses of equalization is at a live event, where microphones and

speakers operate simultaneously. An equalizer is used to ensure that there are no

frequency bands where there is a round trip gain of greater than 1, as these are heard as

audible feedback. Those frequencies are cut at the equalizer to prevent this.

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Early telephone systems used equalization to correct for the reduced level of high

frequencies in long cables. For example, a particular microphone might be more sensitive

to low frequency sounds than to high frequency sounds, so an equalizer would be used to

increase the volume of the higher frequencies (boost), and reduce the volume of the low

frequency sounds (cut). Modern systems have less trouble in the voice frequency range,

but DSL circuits operating in the MHz range on those same wires may suffer severe

attenuation distortion which is dealt with by automatic equalization or by abandoning the

worst frequencies. Picture phone circuits also had equalizers.

adjacent telephone wires. This occurs due to inductance, capacitance or electrical

conductance coupling. The higher frequencies are more likely to transfer to adjacent

wires. The band pass that the phone company allows through is 400 through 3,400 Hz.

Any frequencies above or below this are filtered out. These frequencies are acceptable for

voice speech to be transmitted clearly.

After equalization, the effect of the channel's low-pass filter is removed, but the

additive noise is still there. It manifests itself as causing the received constellation points

to deviate from their location in the original constellation. To enable the bit stream to be

recovered, a nearest-neighbor approximation is performed on each point. As long as the

noise amplitude is small or the constellation points are far apart, it is unlikely that any

single point will deviate enough from it's original location such that it has a new nearest-

neighbor. With high noise power, however, the points are scattered all over the

constellation; the nearest neighbor in this case is unlikely to be the original point. In our

system, we implemented this approximation with a parser and look up table; we would

examine each complex value.

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CHAPTER III

Software Environment:

Hardware Environment:

Processor : Pentium - II

RAM : 128MB

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Introduction

Matlab is a tool for doing numerical computations with matrices and vectors. It can

also display information graphically. The best way to learn what Matlab can do is to work

through some examples at the computer. MATLAB is a high performance language for

to use environment .Mat lab stands for matrix laboratory. It was written originally to

provide easy access to matrix software developed by LINPACK (linear system package)

foundation of sophisticated matrix software in which the basic element is matrix that does

2. Algorithm development

3. Data acquisition

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individual applications.

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MATLAB

MATLAB

Programming language

Computation

Graphics

Linear algebra External interface

2-D graphics

Signal processing Interface with C and

3-D graphics

Quadrature FORTRAN

Color and lighting

Etc Programs

Animation

Tool boxes

Signal processing

Image processing

Control systems

Neural Networks

Communications

Robust control

Statistics

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Development Environment.

This is the set of tools and facilities that help you use MATLAB functions and files.

Many of these tools are graphical user interfaces. It includes the MATLAB desktop and

Command Window, a command history, an editor and debugger, and browsers for

functions, like sum, sine, cosine, and complex arithmetic, to more sophisticated functions

like matrix inverse, matrix Eigen values, Bessel functions, and fast Fourier transforms.

"programming in the small" to rapidly create quick and dirty throw-away programs, and

Graphics.

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MATLAB has extensive facilities for displaying vectors and matrices as graphs, as

well as annotating and printing these graphs. It includes high-level functions for two-

presentation graphics. It also includes low-level functions that allow you to fully

This is a library that allows you to write C and Fortran programs that interact with

MATLAB. It includes facilities for calling routines from MATLAB (dynamic linking),

calling MATLAB as a computational engine, and for reading and writing MAT-files.

3.5Starting MATLAB

icon on your Windows desktop. On UNIX platforms, start MATLAB by typing mat lab

at the operating system prompt. You can customize MATLAB startup. For example, you

can change the directory in which MATLAB starts or automatically execute MATLAB

The answer to the most popular question concerning any program is this: leave a

quit

or by typing

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MATLAB Desktop

When you start MATLAB, the MATLAB desktop appears, containing tools (graphical

user interfaces) for managing files, variables, and applications associated with MATLAB.

The following illustration shows the default desktop. You can customize the arrangement

of tools and documents to suit your needs. For more information about the desktop tools .

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3.6 Implementations

1. Arithmetic operations

Entering Matrices

The best way for you to get started with MATLAB is to learn how to handle

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Start by entering Dürer’s matrix as a list of its elements. You only have to

A=

16 3 2 13

5 10 11 8

9 6 7 12

4 15 14 1

This matrix matches the numbers in the engraving. Once you have entered

Functions:

There are many functions which we apply to scalars which Matlab can apply to both

sin(d)

exp(d)

log(d)

abs(d)

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Matlab has functions to round floating point numbers to integers. These are round, fix,

ceil, and floor. The next few examples work through this set of commands and a couple

more arithmetic operations.

f=[-.5 .1 .5]

round(f)

fix(f)

ceil(f)

floor(f)

sum(f)

prod(f)

Relations and Logical Operations

In this section you should think of 1 as "true" and 0 as "false." The notations &, |,

~ stand for "and,""or," and "not," respectively. The notation == is a check for equality.

a=[1 0 1 0]

b=[1 1 0 0]

a==b

a<=b

~a

a&b

a & ~a

a|b

a | ~a

M-Files

You can create your own matrices using M-files, which are text files containing

MATLAB code. Use the MATLAB Editor or another text editor to create a file with the

same statements you would type at the MATLAB command Line. Save the file under a

name that ends in .m.

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Graph Components

you need to define a coordinate system. Therefore every graph is placed within axes,

which are contained by the figure. The actual visual representation of the data is achieved

with graphics objects like lines and surfaces. These objects are drawn within the

coordinate system defined by the axes, which MATLAB automatically creates

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specifically to accommodate the range of the data. The actual data is stored as properties

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Plotting Tools

Plotting tools are attached to figures and create an environment for creating

Display the plotting tools from the View menu or by clicking the plotting tools

Editor/Debugger

Use the Editor/Debugger to create and debug M-files, which are programs you write

to run MATLAB functions. The Editor/Debugger provides a graphical user interface for

text editing, as well as for M-file debugging. To create or edit an M-file use File > New

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simulink and communication block set and mathematical block set in MATLAB

Elementary matrices.

zeros - Zeros matrix.

ones - Ones matrix.

eye - Identity matrix.

rand - Uniformly distributed random numbers.

randn - Normally distributed random numbers.

linspace - Linearly spaced vector.

Trigonometric.

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sin - Sine.

sinh - Hyperbolic sine.

asin - Inverse sine.

asinh - Inverse hyperbolic sine.

cos - Cosine.

cosh - Hyperbolic cosine.

acos - Inverse cosine.

acosh - Inverse hyperbolic cosine.

tan - Tangent.

tanh - Hyperbolic tangent.

atan - Inverse tangent.

atan2 - Four quadrant inverse tangent.

atanh - Inverse hyperbolic tangent.

sec - Secant.

sech - Hyperbolic secant.

asec - Inverse secant.

asech - Inverse hyperbolic secant.

csc - Cosecant.

csch - Hyperbolic cosecant.

acsc - Inverse cosecant.

acsch - Inverse hyperbolic cosecant.

cot - Cotangent.

coth - Hyperbolic cotangent.

acot - Inverse cotangent.

acoth - Inverse hyperbolic cotangent.

Exponential.

exp - Exponential.

log - Natural logarithm.

log10 - Common logarithm.

sqrt - Square root.

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Complex.

abs - Absolute value.

angle - Phase angle.

conj - Complex conjugate.

imag - Complex imaginary part.

real - Complex real part.

help - On-line documentation.

lookfor - Keyword search through the HELP entries.

demo - Run demos.

who - List current variables.

whos - List current variables, long form.

load - Retrieve variables from disk.

Polynomials.

roots - Find polynomial roots.

poly - Construct polynomial with specified roots.

polyval - Evaluate polynomial.

polyvalm - Evaluate polynomial with matrix argument.

residue - Partial-fraction expansion (residues).

polyfit - Fit polynomial to data.

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conv - Multiply polynomials.

deconv - Divide polynomials.

+ Plus arith

- Minus arith

* Matrix multiplication arith

.* Array multiplication arith

^ Matrix power arith

.^ Array power arith

\ Backslash or left division slash

/ Slash or right division slash

./ Array division slash

: Colon colon

() Parentheses paren

[] Brackets paren

. Decimal point punct

, Comma punct

; Semicolon punct

% Comment punct

' Transpose and quote punct

= Assignment punct

== Equality relop

<,> Relational operators relop

& Logical AND relop

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| Logical OR relop

~ Logical NOT relop

xor Logical EXCLUSIVE OR xor

plot - Linear plot.

loglog - Log-log scale plot.

semilogx - Semi-log scale plot.

semilogy - Semi-log scale plot.

fill - Draw filled 2-D polygons.

Graph annotation.

title - Graph title.

xlabel - X-axis label.

ylabel - Y-axis label.

zlabel - Z-axis label for 3-D plots.

grid - Grid lines.

axis - Axis scaling and appearance.

Loops

0.8 0.1

0.2 0.9

and let x be the column vector

1

0

We regard x as representing (for example) the population state of an island. The first

entry (1) gives the fraction of the population in the west half of the island, the second

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entry (0) give the fraction in the east half. The state of the population T units of time later

is given by the rule y = ax. This expresses the fact that an individual in the west half stays

put with probability 0.8 and moves east with probability 0.2 (note 0.8 + 0.2 = 1), and the

fact that in individual in the east stays put with probability 0.9 and moves west with

repeated matrix multiplication. This can be done by the following Matlab program:

>> x = [ 1; 0 ]

>> for i = 1:20, x = a*x, end

This has been a sample of the basic MATLAB functions and the matrix

manipulation techniques. At the end of the tutorial there is a listing of functions. The

functions that you have available will vary slightly from version to version of MATLAB.

By typing help

• Advantages:

Handles vector and matrices very nice

Quick plotting and analysis

EXTENSIVE documentation (type ‘help’)

Lots of nice functions: FFT, fuzzy logic, neural nets, numerical integration,

OpenGL

• Drawbacks:

Slow compared to C or Java

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CHAPTER IV

EXPERIMENTAL RESULTS

Implementation:

Implementation is the stage where the theoretical is turned into a working system.

The most crucial stage in achieving a new successful system and in giving confidence on

the new system for the users that it will efficiently and effectively.

The system can be implemented only after thorough testing is done and if it is found

It involves careful planning, investigation of the current system and its constraints on

change over methods a part from planning. Two major tasks of preparing the

implementations are education and training of the users and testing of the system.

The more complex the system being implemented the more involved will be the

The implementation phase comprises of several activities. The required hardware and

software acquisition is a carried out. The system may require some software to develop.

For this, programs are written and tested. The user then changes over to his new fully

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Testing:

finding errors and missing operations and also a complete verification to determine

whether the objectives are met and the user requirements are satisfied.

The first includes unit testing, where in each module is tested to provide its

correctness, validity and also determine any missing operation and to verify whether the

objectives have been met. Errors are noted down and corrected immediately. Unit

testing is the important and major part of the project. So errors are rectified easily in

particular module and program clarity is increased. In this project entire system is

divided into several modules and is developed individually. So unit testing is conducted

to individual modules.

The second step includes integration testing. It need not be the case, the software

whose modules when run individually and showing perfect results, will also show perfect

results when run as a whole. The individual modules are clipped under this major

module and tested again and verified the results. This is due to poor interfacing, which

may result in data being lost across an interface. A module can have inadvertent, adverse

effect on any other or on the global data structures, causing serious problems.

The final step involves validation and testing which determines which the software

functions as the user expected. Here also some modifications were. In the completion of

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4.2.3 NOISE :

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CHAPTER V

CONCLUSION

The goal of this project is to give a little insight into the adaptive filter theory and

show some basic aspects in noise cancellation in speech enhancement. Understanding the

filter structure and the properties of each adaptive algorithm plays a key role in the

process of implementation on DSP processors. The main impact is cast on cost reduction

of the adaptive systems - final products - as well as retaining the high quality of noise

suppression. As mentioned in the introduction, when designing an adaptive system, it is

necessary to accept a number of trade-offs between disparate requirements, which cannot

be satisfied at the same time. For example the LMS-family algorithms are very simple,

tractable and computationally efficient. But their essential disadvantage is the slow

convergence rate and bigger steady-state error.

On the other one side, RLS-family algorithms are computationally more complex and

also structurally complicated. Here, the proper set-up and tuning of system parameters

requires deeper experience in the domain of adaptive filtration. However, a great

contribution can be addressed to precise adaptive mechanism with a low steady-state

error and extremely high convergence rate. In the future we can expect that the increasing

performance of DSP processors as well as the quality of DSP services will cause

requirements such as computational modesty or memory consumption irrelevant. Still

there is a fact that higher standard does not come for nothing and the price as we know is

a key factor for majority of customers. Finally, adaptive noise cancellation in speech

signals is a huge domain of research, full of opportunities and unsolved problems. It is

definitely worth to spend the future time in its exploration.

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“Filtered” LMS problems. This new approach formulates the adaptive filtering (control)

problem as an H estimation problem, and updates the adaptive weight vector according

to the state estimates provided by an H estimator. This estimator is proved to be always

feasible.

algorithm) has provable performance, follows a simple update rule, and unlike previous

methods readily extends to multi-channel systems.

The proposed method shows much faster convergence and improved steady-state

performance. Conventional methods can go unstable depending on adaptation parameter

or step-size (µ).

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BIBLIOGRAPHY

Sept.1985.

S. Haykin, Adaptive Filter Theory. 2nd Edition, Prentice Hall: Englewood Cli_s,

1991. Chapter13.

J.G.Provakis,Digital Communications,Newyork:TataMcGraw-Hill,1985.

E.H. Anderson and J.P. How. Active Vibration Isolation Using Adaptive Feed-

forward Control. ACC97 , pages 1783–1788, July 1997.

M.R. Bai and Z. Lin. Active Noise Cancellation for a Three-Dimensional En-

closure by Using Multiple Channel Adaptive Control and H Control. ASME

Journal of Vibration and Acoustics , 120:958–964, o ctober 1998.

D.S. Bernstein and W.M. Haddad. LQG Control with anH PerformanceBound: A

Riccati Equation Approach. IEEE Trans. on Auto. Control , 34:293–305, 1989.

N.J. Bershad, P.L. Feintuch, F.A. Reed, and B. Fisher. Tracking Characteristicsof

the LMS Adaptive LineEnhancer-Response to a Linear Chirp Signal in

Noise.IEEE Trans. on Acoust., Speech, Signal Processing, 28:504–516, October

1980.

Noise, Part 2: Performance of the LMS AlgorithmIEEE Trans. on

SignalProcessing , 39:595–602, March 1991.

Speechand Audio Processing , 3:504–514, November 1995.

MatrixInequalities in System and Control Theory . SIAM, 1994.

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WEBSITES

www.google.com

www.eece.unm.edu

www.mathworks.com

www.IEEE.com

www.ece.utah.edu

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