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Transfer Function Digital Filters

An adaptive filter is a filter that can automatically adjust its transfer function based on an optimization algorithm and error signal to match changing parameters in an application. It is needed when the parameters of the desired processing are unknown beforehand. The adaptive process involves minimizing a cost function to modify the filter transfer function on each iteration based on input and an error feedback signal. Common applications of adaptive filters include noise cancellation, signal prediction, echo cancellation, and adaptive feedback cancellation.

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79 views4 pages

Transfer Function Digital Filters

An adaptive filter is a filter that can automatically adjust its transfer function based on an optimization algorithm and error signal to match changing parameters in an application. It is needed when the parameters of the desired processing are unknown beforehand. The adaptive process involves minimizing a cost function to modify the filter transfer function on each iteration based on input and an error feedback signal. Common applications of adaptive filters include noise cancellation, signal prediction, echo cancellation, and adaptive feedback cancellation.

Uploaded by

charanf22
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as DOCX, PDF, TXT or read online on Scribd
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An 

adaptive filter is a filter that self-adjusts its transfer


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

1 Example application

2 Block diagram

3 Applications of adaptive

filters

4 Filter implementations

5 See also

6 References

[edit]Example application
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). However, due to slight variations in the power
supply to the hospital, the noise signal may contain harmonics of the
noise and the exact frequency of the noise may vary.
One way to remove the noise is to filter the signal with a notch filter at 50
Hz. Such a static filter would need to remove all the frequencies in the
vicinity of 50 Hz, which could excessively degrade the quality of the ECG
since the heart beat would also likely have frequency components in the
rejected range.
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.
[edit]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 we 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.
[edit]Applications of adaptive filters

 Noise cancellation
 Signal prediction
 Adaptive feedback cancellation
 Echo cancellation
[edit]Filter implementations

 Least mean squares filter


 Recursive least squares filter
 Multidelay block frequency domain adaptive
filter
[edit]

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