DSP Test II

1. Using BZT, obtain the discrete transfer difference function of the second order filter
shown in Figure 2a below. Express this as a difference equation suitable for computer
processing. (10)

Figure 2a An LCR filter


Using the BZT substitution, we obtain:

Using the identities:

And substituting into Equation we obtain:

Once again, the objective of the algebraic manipulation must be to express the transfer function as the
ratio of two polynomials in z. Multiplication of both the numerator and denominator of Equation by (1 +
z
-1
)
2
and grouping the z terms produces:

For the purposes of filter gain normalisation, the constant term in the denominator must be unity; to
simplify the algebra, we make use of the following identities:

Equation therefore reduces to:

Finally, the difference equation of the IIR filter is obtained directly from Equation, i.e.


2. Describe the three main ways of designing a digital IIR filters. (10)
Direct digital synthesis, e.g. pole-zero placement.
Impulse invariance
Conversion of analogue filters into their discrete equivalents, e.g. the bilinear z -transform, or
BZT method.

Impulse Invariance
Direct digital IIR filter design is rarely used, for one very simple reason: nobody knows
how to do it. While it is easy to calculate the filter's frequency response, given the filter
coefficients, the inverse problem - calculating the filter coefficients from the desired
frequency response - is so far an insoluble problem. Because we do not know how to
design digital IIR filters, we have to fall back on analogue filter designs (for which the
mathematics is well understood) and then transform these designs into the sampled data
z- plane Argand diagram. Note that the filter's impulse response defines it just as well as
does its frequency response. Here is a recipe for designing an IIR digital filter:

Decide upon the desired frequency response
Design an appropriate analogue filter
Calculate the impulse response of this analogue filter
Sample the analogue filter's impulse response
Use the result as the filter coefficients



This process is called the method of impulse invariance. The method of impulse
invariance seems simple: but it is complicated by all the problems inherent in dealing with
sampled data systems. In particular the method is subject to problems of aliasing and
frequency resolution.

The bilinear z-transform
The BZT allows us to convert an entire analogue filter transfer function based on the
Laplace transform into a discrete transfer function based on the z-transform, and is
perhaps the most important re-mapping technique for the design of digital filters. It works
as follows: wherever s appears in the transfer function, it is replaced by the expression:

3. Briefly describe the two main ways of designing a digital FIR filter indicating which
method would be desirable for linear system emulation. (9)
a. The window method:
b. 1. Using the appropriate formulae listed, calculate the coefficients for the specified filter type.
c. 2. Apply a suitable window function, taking into account the stop band attenuation.
d. 3. Determine the number of coefficients required for a given window type, according to
thedesired transition width.
e. The frequency sampling method. linear system emulation implementation.
f.
g. 1. The frequency response of the filter is specified in the Fourier-domain. For a linear phase
filter, setting the real terms to their intended values, and leaving the imaginary terms as zero
determine the frequency response.
h. 2. The inverse Fourier transform is taken of the designated frequency response. This results in
a time domain signal which is not centred, i.e. the right-hand part of the signal starts at t =0,
and the left-hand part extends backwards from the final value. Hence manipulation is
necessary to centre the impulse response. Once centred, it will appear typically as in Figure b.
i. 3. The ends of the impulse response must now be tapered to zero using a suitable window
function. An impulse response thus modified is shown in Figure c. Application of the window
minimises ripples in the pass and stop bands but it also increases the width of the transition
zone, resulting in a frequency response shown in Figure d.

4. You have been given an out of focus noisy image and using the principles you have learnt
so far in DSP, explain how you would process it using block diagrams wherever possible.
(6)

Can Use Multi-Rate Data Conversion














5. Compare and contrast the properties of FIR and IIR filters (8)

6. A digital filter is described by the expression:
y
n
= 2x
n
- x
n-1
+ 0.8y
n-1
Rectangular pulse LP filter
kernel
Cut-off
Frequency
6Mhz
Sampling
Frequency
12Mhz
S/N 6dB per
bit of required
Resolution
State whether the filter is recursive or non-recursive. Justify your answer. (2)
Recursive: it contains a previous output term, y
n-1



END