DIGITAL FILTERS

A Case study
Digital Filters
 A digital filter is a system that performs mathematical operations on
a sampled, discrete-time signal to reduce or enhance certain aspects of that
signal.
 A digital filter system usually consists of an analog-to-digital converter (ADC)
to sample the input signal, followed by a microprocessor and some peripheral
components such as memory to store data and filter coefficients etc. Program
Instructions (software) running on the microprocessor implement the digital
filter by performing the necessary mathematical operations on the numbers
received from the ADC.
 In some high performance applications, an FPGA(Field-Programmable Gate
Array) or ASIC(Application-Scientific Integrated Circuit) is used instead of a
general purpose microprocessor, or a specialized digital signal processor (DSP)
with specific paralleled architecture for expediting operations such as
filtering.
About the case study

 The purpose of this case study is to present a development of the step-

invariant approach to highpass digital filter design .
 I will try to show that the transfer function, G(z)sif , obtained via the step
invariant design method, may be made to approximate closely to the transfer
function, G(z) obtained via the bilinear z-transform method.
 Also I will try to show that it is possible to convert a time domain (step-
invariant) filter, G(z)si , to one that satisfies a frequency domain specification
,G(z)sif . This can be achieved by observing certain conditions and by
employing a suitable gain term. The validity of the method is demonstrated
using a practical example of a simple highpass filter and a digital phase-
advance network.
The approach

 As we know that the impulse-invariant design method for bandlimited filters, is

based on the application of standard z-transforms, whereby an analogue filter
transfer function, G(s), is transformed to an equivalent digital filter transfer
function ,G(z), that is

 For the analogue filter, the impulse response, g(t), is defined as L-1[G(s)]
Similarly, for a digital filter, the impulse response, g k, is defined as Z-1 [G(z)].
To be impulse-invariant gk= g(t) for t = 0, T, 27, ..., where T is the sampling
period. Furthermore, the frequency response of the digital filter ,G(exp(jωT)),
will approximate to the frequency response of the analogue filter,G(jω), if
aliasing errors have been minimised by band-limiting [G(s)] and by correct
The approach(continued…)

 A common approach to the design of non-bandlimited digital filters (highpass

or bandstop) is to use the well-known bilinear z-transform, that is, by directly
substituting 2 / T[(z -1) / (z + 1)] for s in [G(s)] ,a corresponding transfer
function G(z), is obtained.
 However, non-linear distortion (warping) may be introduced into the filter
representation because of the non-linear relationship between the analogue
filter frequency scale and the digital filter frequency scale. This problem is
resolved by using prewarping techniques.
 An alternative approach to the design of non-bandlimited filters is to use the
step-invariant method described in this case study
HIGHPASS FILTER DESIGN

 A simple analogue highpass filter has a transfer function : G(s) = s /(s + α), where ω = α
rad/ s is the cut-off frequency of the filter. This is non-bandlimited and consequently
the impulse-invariant design method is excluded, however, employing the suitable
bilinear z-transform we obtain

 Referring to this equation it is seen that G(z)bl has a gain term equal to 1/(1+ αT/2),
and the corresponding z-plane representation has a zero at z = 1 and a pole at z =1-
αT/2/(1+ αT/2) that is, the pole is at z = (J - αT + α2T2 /2 – α3T3 /4 + …)
HIGHPASS FILTER DESIGN(continued…)

 Now consider where Y(s) is the Laplace transform of the filter

response and X(s) is the Laplace transform of the filter input signal. For the step-
invariant design method the step input signal is assumed to have an amplitude of A
for t = > 0, i.e. X(s)= A/s therefore Y(s) = (A/s)*(s/s+α) = A/(s+α).
 Taking the inverse Laplace transform of equation above we obtain the corresponding
step response of the analogue filter, thus [Y(s)] = L-1 [Y(s)] = Aexp(-αt) and
Transforming equation mentioned above into the z-domain (standard z-transform)
yields Y(Z) = A*z/(z - exp(- αt)).
 The standard z-transform of X(s)= AI s is X(z) = A*z I ( z - 1), therefore the
corresponding transfer function of the digital filter is
HIGHPASS FILTER DESIGN(continued…)

 Taking the inverse z-transform of equation mentioned in previous slide we obtain the
corresponding step response of the digital filter, thus yk = Z-1 [Y(z)]= Aexp(- αkT).

 Comparing equations for y(t) and yk we see that yk is equal to y(t) at the sampling
instants, and therefore the digital filter defined by equation for G(z) in previous slide
is step-invariant.
 Now returning to the bilinear z-transform of the analogue filter (equation for G(z) bi)
we allow for prewarping by making the substitution