TERM PAPER

OF

DIGITAL SIGNAL PROCESSING

OBJECTIVES:

1 DIFFERENCE BETWEEN SINGLE MODEL AND MULTIMODEL

2 ADAPTIVE FILTER

3 COMPARISION OF LMS AND RLS

4 COMPARISON B/W NLMS AND VLMS

5 COMPARISON B/W VLMS AND BLMS

SUBMITTED BY:

Navpreet Singh

Roll no: 20
Adaptive digital filters

Introduction
An adaptive filter is essentially a digital filter with self- adjusting characteristics. It adapts,
automatically, to changes in its input signals. Adaptive filters are the central topic in the sub-
area of DSP known as adaptive signal processing.LMS(Least Mean Square) and RLS
(recursive least squares) algorithms which are two of the most widely used algorithms in
adaptive signal processing. It has number of real- world application.

An adaptive filter is a filter that self-adjusts its transfer function according to an 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).

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 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.

When and Where to use Adaptive filters

The contamination of signal of interest by other unwanted, often larger , signals or noise is
problem often encountered in many applications . Where the signal and noise occupy fixed
and separate frequency bands, conventional linear filters with fixed coefficients are normally
used to extract the signal . However, there are many instances when it is necessary for filter
characteristics to be variable, adapted to changing signal characteristics, or to be altered
intelligently. In such cases, the coefficients of the filter must vary and cannot be specified in
advance.
Such is the cases, the coefficients of a spectral overlap between the signals and noise as
shown in figure or if varies with time Typical applications where fixed coefficient filters
are inappropriate are the following:

(1) Electroencephalography (EEG), where artefacts or signal contamination produced by

eye movements or blinks is much larger than the genuine electrical activity of the
brain and shares the same frequency band with signals of clinical interest. It is not
possible to use conventional linear filters to remove the artefacts while preserving the
signals of clinical interest.
(2) Digital communication using spread spectrum, where a larger jamming signals,
possibly indented to disrupt communication , could interfere with the desired signal.
The interference often occupies a narrow but unknown band within the wideband
spectrum, and can only be effectively dealt with adaptively.
(3) In digital data communication over the telephone channel at a high rate. Signal
distortions caused by the poor amplitude and phase response characterstics of channel
lead to pulses representating different digital codes to interfere with each other (inter
symbol interference), making it difficult digital codes to detect the codes reliably at
the receiving end .To compensate for the channel distortion s which may be varying
with time or of unknown characteristics at the receiving end adaptive equalization is
used.
An adaptive filter has property that its frequency response is adjustable or modificable
automatically to improve its performance in accordance with some criteria, allowing the filter
to adapt to changes in the input signal characterstics. Because of their self – adjusting
performance and in built flexibility, adaptive filters have found use in many diverse
applications such as telephone echo cancelling, radar signal processing, navigational systems
equalization of communication channels, and biomedical signal enhancement.

In summary adaptive filters are used:

 When it is necessary for the filter characterstics to be variable, adapted to changing

conditions;
 When there is spectral overlap between the signal and noise
 If the band occupied by the noise is unknown or varies with time.

The use of conventional filters in above cases would lead to unacceptable distortion of the
desired signal. There are many other situations, apart from noise reduction, when the use of
adaptive filters is appropriate.

Block diagram

The block diagram, shown in the following figure, serves as a foundation for particular
adaptive filter realisations, such as Least Mean Squares (LMS) and Recursive Least Squares
(RLS). The idea behind the block diagram is that a variable filter extracts an estimate of the
desired signal.

To start the discussion of the block diagram it take the following assumptions:

 The input signal is the sum of a desired signal d(n) and interfering noise v(n)
 x(n) = d(n) + v(n)

 The variable filter has a Finite Impulse Response (FIR) structure. For such structures
the impulse response is equal to the filter coefficients. The coefficients for a filter of
order p are defined as

 The error signal or cost function is the difference between the desired and the
estimated signal

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

where

is an input signal vector. Moreover, the variable filter updates the filter coefficients at every
time instant

where is a correction factor for the filter coefficients. The adaptive algorithm generates
this correction factor based on the input and error signals. LMS and RLS define two different
coefficient update algorithms.

Concepts of adaptive filtering

Adaptive filters as a noise canceller

An adaptive filter consists of two distinct parts:

1. A digital filter with adjustable coefficient.

2. An adaptive algorithm which is used to adjust or modify the coefficients of the filter.

The two input signals, yk and xk , are applied simultaneously to the adaptive filter. The signal
yk is the contaminated signal combining both the desired signal, sk, and the noise nk , assumed
uncorrelated with each other. The signal, xk is a measure of the contaminating signal which is
ˆ
correlated in some way with nk. xk is processed by the digital filter to produce an estimate, n k
of nk.. An estimate of the desired signal is then obtained by subtracting the digital filter
output, nˆ , from the contaiminated signal, yk:

sˆk = yk - nˆk = sk + nk - nˆk

Block diagram of an adaptive filter as a noise canceller

The main objective in noise cancelling is to produce an optimum estimate of noise in the
contaminated signals and hence an optimum estimate of the desired signal. This is achived by
using sˆk in a feedback arrangement to adjust the digital filter coefficients, via a suitable
adaptive algorithm , to minimize the noise in sˆk. the output signal, sˆk ,serves two purposes;
 (i) An estimate of the desired signal
(ii) Error signal which is used to adjust the filter coefficients.

Different configurations of adaptive filters

Except adaptive filters used in noise cancellation it can be used for other purposes also, such
as for linear prediction , adapative signal enhancement and adaptive control. In general, the
meaning of the signals xk,yk,and ek or the way they are derived are application dependent,
afact which should be borne in mind, figure shows different configurations of adaptive filter.

Some configurations of adaptive filters (a) adaptive noise canceller;(b) adaptive self
tunning filter (c) cancelling periodic interference without an external refrence source
(d) adaptive linear enhancer

Main Components of adaptive filter

In most adaptive systems, the digital filter is relaized using FIR structure. Other forms are
sometimes used, for example the infinite impulse response (IIR) or the lattice structures, but
the FIR structure is the most widely used because of its simplicity and guaranteed stability.
For the N-point filter depicted in figure , the output is given by

Where wk(i),i= 0,1,……., are the adjustable filter coefficients (or weights) and xk(i) and nˆk
are the input and output of the filter. Figure shows single-input, single- output system. In a
multiple- input single – output system, the xk may be simultaneous inputs from N different
signal sources.

Derivation of the LMS algorithm

By relaxing the infinite sum of the Wiener filter to just the error at time n, the LMS algorithm
can be derived.

The squared error can be expressed as:

Using the Minimum mean-square error criterion, take the gradient:

Apply chain rule and substitute definition of

From this equation, we can derive an update equation for each hi at every new n using

gradient descent and a step size μ:

which becomes, for i = 0, 1, ..., N-1,

This is the LMS update equation.

Adaptive algorithms

Basic Weiner filter theory

Basic LMS Algorithm

Implementation of LMS (flow Chart)

Hardware Implementation

Limitations of LMS

Types of LMS

RLS Algorithm

RLS limitation

Applications of Adaptive filter

1. Electroencephalography (EEG)
2. Real time Implementation
3. Artefact Processing
4. Adaptive Telehone echo cancellation
5. Multipath Compensation
6. Adaptive Jammer Supression
7. Radar Signal Processing
8. Fetal Monitoring detail study