Digital Signal Processing
Digital Signal Processing
UNIT V
Digital filters Design Techniques
Design of IIR and FIR digital filters, Impulse invariant and bilinear transformation, windowing techniques
rectangular and other windows, examples of FIR filters, design using windowing.
FIR FILTER
FIR system has finite duration unit sample response. i.e. h = for < a d M. Thus the unit sample
response exists for the duration from 0 to M-1. FIR systems are non recursive. Thus output of FIR filter
depends upon present and past inputs.
Difference equation of the LSI system for FIR filters becomes
M
y(n)= ∑ bk x(n–k)
k=0
FIR systems has limited or finite memory requirements. FIR filters are always stable FIR filters can have an
exactly linear phase response so that no phase distortion is introduced in the signal by the filter. The effect of
using finite word length to implement filter, noise and quantization errors are less severe in FIR than in IIR.
FIR filters can have an exactly linear phase response so that no phase distortion is introduced in the signal by
the filter.
The effect of using finite word length to implement filter, noise and quantization errors are less severe in FIR
than in IIR. FIR filters are generally used if no phase distortion is desired.
IIR FILTER:
IIR system has infinite duration unit sample response. i. e h(n) = 0 for n<0 Thus the unit sample response exists
for the duratio fro to ∞. IIR systems are recursive. Thus they use feedback. Thus output of IIR filter
depends upon present and past inputs as well as past outputs
Difference equation of the LSI system for IIR filters becomes
N M
y(n)= -∑ ak y(n–k + ∑ bk x(n–k)
k=1 k=0
IIR filter is usually more efficient design in terms of computation time and memory requirements. IIR systems
usually require less processing time and storage as compared with FIR.
IIR filter is usually more efficient design in terms of computation time and memory requirements. IIR systems
usually require less processing time and storage as compared with FIR.
Analogue filters can be easily and readily transformed into equivalent IIR digital filter. But same is not possible
in FIR because that have no analogue counterpart. IIR filters are generally used if sharp cutoff and high
throughput is required.
In impulse variance method, Analog filters are converted into digital filter just by replacing unit sample
response of the digital filter by the sampled version of impulse response of analog filter. Sampled signal is
obtained by putting t=nT hence
where h(n) is the unit sample response of digital filter and T is sampling interval.
But the main disadvantage of this method is that it does not correspond to simple algebraic mapping of S
plane to the Z plane. Thus the mapping from analog frequency to digital frequency is many to one. The
segments (2k- ∏/T Ω k+ ∏/T of jΩ a is are all apped o the u it ir le ∏ ω ∏. This takes pla e
because of sampling.
Frequency aliasing is second disadvantage in this method. Because of frequency aliasing, the frequency
response of the resulting digital filter will not be identical to the original analog frequency response.
Because of these factors, its application is limited to design low frequency filters like LPF or a limited class of
band pass filters.
Z is represented as rejω in polar form and relationship between Z plane and S plane is given as Z=e ST where s=
σ + j Ω.
Z= e (σ + j Ω) T
= eσT . ejΩT
r= e σ T and ω = Ω T
1) If σ = the r=
1 1
1
s-a 1- eaT z-1
Bilinear transformation method (BZT) is a mapping from analog S plane to digital Z plane. This
conversion maps analog poles to digital poles and analog zeros to digital zeros. Thus all poles and zeros
are mapped.
This transformation is basically based on a numerical integration techniques used to simulate an
integrator of analog filter.
There is one to one correspondence between continuous time and discrete time frequency points.
E tire ra ge i Ω is apped o l o e i to the ra ge -∏ ω ∏.
Frequency relationship is non-linear. Frequency warping or frequency compression is due to non-
linearity. Frequency warping means amplitude response of digital filter is expanded at the lower
frequencies and compressed at the higher frequencies in comparison of the analog filter.
But the main disadvantage of frequency warping is that it does change the shape of the desired filter
frequency response. In particular, it changes the shape of the transition bands.
DIFFERENCE BETWEEN IMPULSE INVARIANCE BILINEAR TRANSFORMATION
In this method IIR filters are designed having a This method of IIR filters design is based on the
unit sample response h(n) that is sampled trapezoidal formula for numerical integration.
version of the impulse response of the analog
filter.
In this method small value of T is selected to The bilinear transformation is a conformal mapping
minimize the effect of aliasing. that tra sfor s the j Ω a is i to the unit circle in the z
plane only once, thus avoiding aliasing of frequency
components.
They are generally used for low frequencies For designing of LPF, HPF and almost all types of Band
like design of IIR LPF and a limited class of pass and band stop filters this method is used.
band pass filter
All poles are mapped from the s plane to the z All poles and zeros are mapped.
plane by the relationship
EXAMPLE:
Using bilinear transformation obtain H(z) if
H(s) = 1___
(s+1)2
And T = 0.1s.
SOLUTION:
For Bilinear transformation
H(z) = H(s) where s = 2 (z-1)___
T (z+1)
H(z) = 1_________
2 (z-1) + 1 2
T (z+1)
Substituting T = 0.1 s
Further simplifying
System is stable only if system produces bounded output for every bounded input. This is stability definition
for any system.
Here h (n) = {b0, b1, b2,} of the FIR filter are stable. Thus y (n) is bounded if input x (n) is bounded. This means
FIR system produces bounded output for every bounded input. Hence FIR systems are always stable.
Windowing Technique
Various window functions are used for filter design.
Following are the types of window functions:
Rectangular Window:
1 for |n| M - 1
WR = 2
0 otherwise
Hamming Window:
Hanning Window:
The width of the main lobe is approximately 8π/M a d the peak of the first side lo e is at- 32dB.