Digital Filter Design

Prepared by:
Asmaa Mosbeh

Under supervision of:

Dr Nabil Sabor
What ..?
• A program Instructions (software) running on the processor (PC or
DSP chip) to perform numerical calculations on sampled values of the
signal.
Why..?
• Embedded in a chip.
• Programmable.
• Flexible (easily changed without affecting hardware).
• More stable (not suffer from time and temperature).
• Less complex (roll-off).
• More versatile (adapt to changes in signal).
• Digital systems.
How..?
• Consider the rational system function

• In filter design, we seek to find the system coefficients, i.e. M, N, ak and bk,
such that the corresponding frequency response

• provides a good approximation to a desired response Hd(ω), i.e.

• The resulting system H(z) must be stable and causal, which imposes strict
constraints on the possible values of ak.
• Additionally, a linear phase requirement for H(z) may be desirable in
certain applications, which further restrict the system parameters.

• FIR ("finite impulse response") or IIR ("infinite impulse response").

• Basic design principle:
•If GLP is essential ⇒ FIR
•If not ⇒ IIR usually preferable (can meet specifications with lower
complexity)
• Typical low-pass specifications

The parameters are:

• [0,ωp ] = pass-band
• [ωs ,π] = stop-band
• δω = ωs − ωp = width of transition band.
• δ1 = pass-band ripple. Often expressed in dB via 20log10(1+δ1 ).
• δ2 = stop-band ripple. Usually expressed in dB as 20log10(δ2 ).
Design of FIR filters
• The general FIR system that is

• Digital FIR filters cannot be derived from analog filters, rational analog
filters cannot have a finite impulse response.

• Stability is not an issue since FIR filters are always stable;

• Considerable emphasis is usually given to the issue of linear phase
Linear-phase filters
• A general FIR filter does not have a linear phase response but
this property is satisfied when

• There are 4 types of FIR GLP filters:

Filter Design by Windowing
• Simplest way of designing FIR filters
• Method is all discrete-time no continuous-time involved
• Start with ideal frequency response

• Choose ideal frequency response as desired response

• Most ideal impulse responses are of infinite length
• The easiest way to obtain a causal FIR filter from ideal is
• More generally

Where
Properties of Windows
• Prefer windows that concentrate around DC in frequency
Less smearing, closer approximation
• Prefer window that has minimal span in time
Less coefficient in designed filter, computationally efficient
• So we want concentration in time and in frequency
Contradictory requirements
• Example: Rectangular window
•U
Conflicting Ideal Requirements
• Use windows with no abrupt discontinuity in their time domain response
and consequently low side-lobes in their frequency response.

• In this case, the reduced ripple comes at the expense of a wider transition
region but this However, this can be compensated for by increasing the
length of the filter.

• Windows with no abrupt discontinuity can be used to reduce Gibbs

oscillations (e.g. Hanning, Hamming, Blackman).
We see clearly that a wider transition region (wider main-lobe) is compensated by
much lower side-lobes and thus less ripples.
Kaiser Window
• Parameterized equation forming a set of windows
Parameter to change main-lob width and side-lob area trade-off
Kaiser Window Parameters

• Given the peak approximation error or in dB as

• and transition band width
• The shape parameter B should be

• The filter order M is determined approximately by

Example
• Design a type I low-pass filter according to the specification
The band-edge frequency of the ideal response if the midpoint between ωs
and ωp
ωc = (ωs + ωp)/2 = (0.2π+0.3π)/2 = 0.25π

• FIR Design by Optimization

Least-Square Method.
Equiripple Design.
Remez method ..etc.
**Recall Our Example:
a least-square method design gives N =
33 (compared to N = 80).
Frequency response of various filters
MATLAB Prototype Filter Design Commands
• [B,A] = BUTTER(N,Wn)
• [B,A] = CHEBY1(N,R,Wn)
• [B,A] = CHEBY2(N,R,Wn)
• [B,A] = ELLIP(N,Rp,Rs,Wn)
– N = filter order
– R = pass band ripple (cheby1) or stop-band ripple (cheby2) in dB. (Rp and Rs respectively for the
elliptic filter)
– Wn = cut-off frequency (normalized)
– [B,A] = filter coefficients.
• C=Fir1(N,Wn,’ftype’) %’ftype’ not needed if it LP filter
For more details please check help tool in MATLAB
• fdatool %in MATLAB command window

➢ In command window, FDAtool will be opened.There you can select FIR or IIR
filter, order of filter and cutoff frequency of a filter (either HPF, LPF or BPF). That
code will automatically generate .m file for you.
Highpass Filter
 Design of IIR filters
For H(z) to be the system function of a causal & stable
LTI system, all its poles have to be inside the unit circle (U.C.).

•Design problem

The goal is to find filter parameters N,M,{ak} and {bk} such that H(ω) ∼ Hd (ω) to
some desired degree of accuracy.
•Transformation methods:

There exists a huge literature on analog filter design Ha(Ω). The transformation methods
try to take advantage of this literature in the following way:

1.Map DT specifications Hd(ω) into analog specifications Had(Ω) via proper

transformation;
2. Design analog filter Ha(Ω) that meets the specifications;
3. Map Ha(Ω) back into H(ω) via a proper inverse transformation.
Impulse invariance method: Design steps
1.Given the desired specifications Hd(ω) for the DT filter, find the corresponding
specifications for the analog filter, Had (Ω); For example: case of a low-pass design

2.Design an analog IIR filter Ha(Ω) that meets the desired analog specifications
Had(Ω):
• Choose the filter type (Butterworth, elliptic, etc.)
• Select the filter parameters (filter order N, poles sk, etc.)

3. Transform the analog filter Ha(Ω) into a discrete-time filter H(z)

via time-domain sampling of the impulse response:
Example: Impulse invariance method
Apply the impulse invariance method to an appropriate analog
Butterworth filter in order to design a low-pass digital filter H(z)
that meets these specifications

• Step 1: We set the sampling period to Ts = 1 (i.e. Ω = ω ), so that the desired specifications for the
analog filter are :

• Step 2: design an analog Butterworth filter Hc(Ω) that meets the above specifications. For the
Butterworth family, we have

• Thus, we need to determine the filter order N and the cut-off frequency Ωc so that the above
inequalities are satisfied.
Step 3: The desired digital filter is finally obtained by applying the
impulse invariance method:
Bilinear transformation
Principle: Suppose we are given an analog IIR filter Ha(s). A digital filter H(z) can be obtained by applying the bilinear
transformation (BT), defined as:
Example: Bilinear transformation
Step 1: Set the parameter α to 1. The desired
specifications for the analog filter are similar to
the above except for the band-edge frequencies.
That is
• Step 2: Design a Butterworth filter Ha(Ω) that meets the above specifications.
• Find that the required order of the Butterworth filter is N = 11 (or precisely N≥ 10.15).
• Determine the cut-off frequency Ωc. For example:

• The desired analog Butterworth filter is specified as

• Step 3: Desired digital filter is obtained by applying BT:

% generate signal
fs = 100; % sampling frequency
f = 5; % signal frequency
t = 5; % time duration
n = [0:1/fs:t]; % sample vector
x = cos (2*pi*f*n);
y=rand(1,length (x)); % y= awgn(x,1.5); “it can also generate noise”
z = x+y;
subplot(3,1,1); plot(n,x); title('Sinusoidal signal');
subplot(3,1,2); plot(n,y); title(‘noise');
subplot(3,1,3); plot(n,z); title('signal with noise');

% IIR Butterworth LPF filter Design

M = 40; % order of the filter
Wc = 2*pi*f/fs; % normalized cut-off frequency (as of signal freq)
[b,a] = butter(M,Wc,'low'); % b,a coeff. of IIR butterworth filter
fvtool(b,a); % filter frequency response MATLAB Ex
% filter the signal
x_f_iir = filter(b,a,z);
figure (3); plot(n,x_f_iir); title('Filtered Sinusoidal Signal with IIR');

% FIR Filter Design

N = 40;
d = fir1(N,Wc); % FIR low pass filter
figure (4); freqz(d,1,500); % frequency response of FIR LPF filter

% filter the signal

x_f_fir = filter(d,1,z);
figure (5); plot(n,x_f_fir); title('Filtered Sinusoidal Signal with FIR');
Thanks