A REPORT ON

ADAPTIVE FILTER AND ITS

APPLICATIONS

SUBMITTED TO:ELECTRONICS AND

COMMUNICATION
DEPARTMENT

PREPARED BY:URJA SHAH

(Me sem 1 CSE)
GUIDE
D BY:-

USHA MAM
ABHAY SIR
Adaptive Filters
The signal degradation in some physical systems is time varying, unknown,
or possibly both. For example, consider a high-speed modem for transmitting
and receiving data over telephone channels. It employs a filter called a
channel equalizer to compensate for the channel distortion. Since the dial-up
communication channels have different and time-varying characteristics on
each connection, the equalizer must be an adaptive filter.
Adaptive filters modify their characteristics to achieve certain objectives by
automatically updating their coefficients. Many adaptive filter structures and
adaptation algorithms have been developed for different applications. This
chapter presents the most widely used adaptive filters based on the FIR filter
with the least-mean-square (LMS) algorithm. These adaptive filters are
relatively simple to design and implement. They are well understood with
regard to stability, convergence speed, steady-state performance, and finiteprecision effects.

Introduction to Adaptive Filtering

An adaptive filter consists of two distinct parts a digital filter to perform the
desired filtering, and an adaptive algorithm to adjust the coefficients (or
weights) of the filter. A general form of adaptive filter is illustrated in Figure
7.1, where d(n) is a desired (or primary input) signal, y(n) is the output of a
digital filter driven by a reference input signal x(n), and an error signal e(n) is
the difference between d(n) and y(n). The adaptive algorithm adjusts the filter
coefficients to minimize the mean-square value of e(n). Therefore, the filter weights
are updated so that the error is progressively minimized on a sample-by sample
basis.

In general, there are two types of digital filters that can be used for adaptive
filtering: FIR and IIR filters. The FIR filter is always stable and can provide a
linear-phase response. On the other hand, the IIR filter involves both zeros
and poles. Unless they are properly controlled, the poles in the filter may
move outside the unit circle and result in an unstable system during the
adaptation of coefficients. Thus, the adaptive FIR filter is widely used for
practical real-time applications. This chapter focuses on the class of adaptive
FIR filters.
The most widely used adaptive FIR filter is depicted in Figure 7.2. The filter
output signal is computed as

where the filter coefficients wl (n) are time varying and updated by the
adaptive algorithms
We define the input vector at time n as

and the weight vector at time n as

y(n) can be expressed in vector form as

The filter output y(n) is compared with the desired d(n) to obtain the error
signal

Our objective is to determine the weight vector w(n) to minimize the

predetermined performance (or cost) function.

Adaptive Noise Cancelation

The widespread use of cellular phones has significantly increased the use of
communication devices in high-noise environments. Intense background
noise, however, often corrupts speech and degrades the performance of
many communication systems. The widely used adaptive noise canceler
employs an adaptive filter with the LMS algorithm to cancel the noise
component embedded in the primary signal.
As illustrated in Figure 7.11, the primary sensor is placed close to the signal
source to pick up the desired signal. The reference sensor is placed close to
the noise source to sense only the noise.
A block diagram of the adaptive noise cancelation system is illustrated in
Figure 7.12, where P(z) represents the transfer function between the noise
source and the primary sensor. The canceler has two inputs: the primary
input d(n) and the reference input x(n). The reference input x(n) contains
noise only. The primary input d(n) consists of signal s(n) plus noise x_(n); i.e.,
d(n) = s(n) + x_(n). The noise x_(n) is highly correlated with x(n) since they
are derived from the same noise source. The objective of the adaptive filter
is to use the reference input x(n) to estimate the noise x_(n). The filter
output y(n), is an estimate of noise x_(n), is then subtracted from the
primary channel signal d(n), producing e(n) as the desired signal plus
reduced noise.
To apply the adaptive noise cancelation effectively, the reference noise
picked up by the reference sensor must be highly correlated with the noise

components in the primary signal. This condition requires a close spacing between
the primary and reference sensors.
Unfortunately, it is also critical to avoid the signal components from the signal
source being picked up by the reference sensor. This crosstalk effect will degrade
the performance of adaptive noise cancelation because the presence of the signal
components in reference signal will cause the adaptive noise cancelation to

cancel the desired signal along with the

undesired noise. Crosstalk problem may be eliminated by placing the primary
sensor far away from the reference sensor.
Unfortunately, this arrangement requires a large-order filter in order to
obtain adequate noise reduction. Furthermore, it is not always feasible to
place the reference sensor far away from the signal Furthermore, it is not
always feasible to place the reference sensor far away from the signal
source.
Second method for reducing crosstalk is to place an acoustic barrier (oxygen
masks used by pilots in aircraft cockpit, for example) between the primary
and reference sensors. However, many applications do not allow an acoustic
barrier between sensors, and a barrier may reduce the correlation of the
noise component in the primary and reference signals. The third technique is
to control the adaptive algorithm to update filter coefficients only during the
silent intervals in the speech. Unfortunately, this method depends on a
reliable speech activity
detector that is very application dependent. This technique also fails to track
the environment changes during the speech periods. In recent years,
microphone array techniques
are used to improve the performance of the noise cancelation.

An example:As shown in Figure 7.12, assume s(n) is a sinewave, x(n) is a white noise, and
P(z) is a simple FIR system. We use the adaptive FIR filter with the LMS
algorithm for noise cancelation.
The adaptive filter will approximate P(z), and thus its output y(n) will
converge to x_(n) in order to cancel it. Therefore, the error signal e(n) will
gradually approach the desired sinewave s(n), as shown in