Professional Documents
Culture Documents
M Tech Microelectronics, Wilp: BITS Pilani
M Tech Microelectronics, Wilp: BITS Pilani
WILP
Prof. Kranthi Kumar Palavalasa.
BITS Pilani Assistant Professor
Pilani Campus Department of EEE (WILP)
BITS Pilani
Pilani Campus
Lecture 13
Implementation of
Discrete-Time Systems
Introduction
Design of IIR filters from analog filters
Frequency transformations
The analog filter having the rational system function H(s), can be described by the linear
constant-coefficient differential equation
linear time-invariant discrete-time systems are also characterized by the rational system
function
Other factors, such as whether the structure or the realization lends itself to parallel
processing, or whether the computations can be pipelined. may play a role in our
selection of the specific implementation.
Furthermore, the unit sample response of the FIR system is identical to the coefficients
{bk},that is,
this structure requires M - 1 memory locations for storing the M - 1 previous inputs,
and has a complexity of M multiplications and M - 1 additions per output point.
For such a system the number of multiplications is reduced from M to M/2 for M
even and to (M- 1)/2 for M odd.
Direct-
form
realization
of linear-
phase FIR
system
Where
Convert the analog band pass filter with system function into a digital IIR filter by use
of the backward difference for the derivative
Convert the analog band pass filter in Example 1 into a digital IIR filter by use of
the mapping
This mapping has introduced two additional poles in the conversion from H a(s) to
H(z). .
As a consequence, the digital filter is significantly more complex than the analog
filter. This is a major drawback to the mapping given above.
Objective is to design an IIR filter having a unit sample response h(n) that is the
sampled version of the impulse response of the analog filter.
Implications:
Continuous time signal xa(t) with spectrum Xa(F) is sampled at a rate Fs = 1/T
samples per second.
The spectrum of the sampled signal is the periodic repetition of the scaled spectrum
FsXa(F)with period Fs.
or, equivalently,
It is also clear that the impulse invariance method is inappropriate for designing high
pass filters due the to spectrum aliasing that results from the sampling process.
Convert the analog filter with system function into a digital IIR filter by means of the
impulse invariance method
Solution We note that the analog filter has a zero at s = -0.1 and a pair of complex
conjugate poles at
A mapping from the s-plane to the z-plane, called the bilinear transformation
overcomes the limitation of the other two design
The BiIinear transformation is a conformal mapping that transforms the jΩ-axis into
the unit circle in the z-plane only once, thus avoiding aliasing of frequency
components.
All points in the LHP of s are mapped inside the unit circle in the z-plane
All points in the RHP of s are mapped into corresponding points outside the unit
circle in the z-plane.
The bilinear transformation can be linked to the trapezoidal formula for numerical
integration.
Integrate the derivative and approximate the integral by the trapezoidal formula.
or, equivalently,
Consequently,
if r < 1, then α < 0: the LHP in s maps into the inside of the unit circle
if r > 1, then α > 0: the RHP in s maps into the outside of the unit circle.
When r =1, then α = 0
Solution
and zeros at
At ω =0, H (0)=1, and at ω =0.2 π we have IH(0.2π)| = 0.707, which is the desired
response.
where
Thus we obtain
It is clear that we must have |g(w)l =1 for all ω.That is, the mapping must be
all-pass. Hence it is of the form