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Digital filters can have characteristic which are not possible with analog filters, such
as truly linear phase response.
Unlike analog filters, the performance of digital filters does not vary with
environmental changes, for example thermal variations. This eliminates the need to
calibrate periodically.
The frequency response of a digital filter can be automatically adjusted if it is
implemented using a programmable processor, which is why they are widely used in
adaptive filters like LMS, NLMS.
Several input signals or channels can be filtered by one digital filter without the
need to replicate the hardware.
Both filtered and unfiltered data can be saved for future use.
With the use of VLSI technology we can fabricate low cost, small size, and low
power digital filter.
Digital filters can achieve thousands of times better performance than analog
filters. This makes a dramatic difference in how filtering problems are approached.
With analog filters, the emphasis is on handling limitations of the electronics, such
as the accuracy and stability of the resistors and capacitors.
Disadvantages of digital filters
Analog filters are cheap and have a large dynamic range in both amplitude and
frequency than digital filters.
The maximum bandwidth of signals that digital filters can handle, in real time,
is much lower than for analog filters. In real time situations, the A/D and D/A
conversion processes introduce a speed constraint on the digital filter
performance. Further, the speed of operation of filter is depends upon the
speed of digital signal processor used and on the number of arithmetic
operations that must be performed for the filtering algorithm, which increases
as the filter response is made tighter.
Digital filters are subjected to ADC noise resulting from quantizing a
continuous signal, and to round off noise incurred during computation. With
higher order recursive filters, the accumulation of round off noise could lead to
instability.
The design and development times for digital filters, especially hardware
development, can be much longer than for analog filters. However, once
developed the hardware and/or software can be used for other filtering or DSP
tasks with little or no modifications
In the design of frequency-selective filters, the
desired filter characteristics are specified in the
frequency domain in terms of the desired
magnitude and phase response of the filter. In the
filter design process, we determine the
coefficients of a causal FIR or IIR filter that
closely approximates the desired frequency
response specifications.
The issue of which type of filter to design, FIR or
IIR , depends on the nature of the problem and on
the specifications of the desired frequency
response.
Types of Digital Filter
1.FIR
2.IIR
In practice, FIR filters are employed in filtering
problems where there is a requirement for a
linear-phase characteristic within the pass band
of the filter.
If there is no requirement for a linear-phase
characteristic, either an IIR or an FIR filter may be
employed. However, as a general rule, an IIR filter
has lower Sidelobes in the stopband than an FIR
filter having the same number of parameters.
Causality and Its Implications
Impulse response h(n) of an ideal low pass filter
with frequency response characteristic:
One possible solution is to introduce a large delay
no in h(n) and arbitrarily to set h(n) = 0 for n < no
However, the resulting system no longer has an
ideal frequency response characteristic. Indeed, if
we set h(n) =0 for n < n0, the Fourier series
expansion of H(w) results in the Gibbs
phenomenon. none of the ideal filter
characteristics are causal, hence all are physically
unrealizable.
h{n) for wc= π/4
What are the necessary and sufficient conditions
that a frequency response characteristic H(w)
must satisfy in order for the resulting filter to be
causal? The answer to this question is given by
the Paley-Wiener theorem