SIGNAL PROCESSING
WITH MATLAB
Presented by:
Farah Hani Nordin
Dr. Farrukh Hafiz
Nagi
What is signal Processing?
The scope of signal processing has grown so broad as to
obviate a perfect and precise definition of what is entailed
in it[1].
Traditionally, signal processing includes the materials
thought in DSP courses but now signal processing has
greater reach because of its influence on related
disciplines such as controls, communications theory and
also digital communication.
Thus, signal processing can be defined as that area of
applied mathematics that deals with operations on or
analysis of signals, in either discrete or continuous time,
to perform useful operations on those signals[1].
What is Signal Processing Toolbox?
The Signal Processing Toolbox is a
collection of tools built on the MATLAB
numeric computing environment. The
toolbox supports a wide range of signal
processing operations, from waveform
generation to filter design and
implementation, parametric modeling, and
spectral analysis. The toolbox provides
two categories of tools.
Command line functions
in the following categories:
Analog and digital filter analysis
Digital filter implementation
FIR and IIR digital filter design
Analog filter design
Filter discretization
Spectral Windows Transforms
Statistical signal processing and spectral analysis
Parametric modeling
Linear Prediction
Waveform generation
A suite of interactive graphical user
interfaces (GUI)for
Filter design and analysis
Window design and analysis
Signal plotting and analysis
Spectral analysis
Filtering signals
Signal Processing Basics
Common Sequences
Since MATLAB is a programming language, an endless variety of
different signals is possible. Here are some statements that generate
several commonly used sequences, including the unit impulse, unit
step, and unit ramp functions:
t = (0:0.01:1);
y = ones(101);
y = [1; zeros(100,1)];
y=t ;
y = t.^2;
y = square(2*pi*4*t);
% step
% impulse
% ramp
% exponential
% generates a square wave every 0.25secs.
Waveform generation
y = sin(2*pi*50*t) + 2*sin(2*pi*120*t); %two sinusoids,
%one at 50 Hz
%and one at
%120Hz with
%twice the amplitude
plot(t,y)
plot(t(1:50),y(1:50))
%plot y versus time
%display only the first
%50 points(zoom!)
Filter Implementation and Analysis
Convolution and Filtering
The mathematical foundation of filtering is convolution. The
MATLAB conv function performs standard one-dimensional
convolution, convolving one vector with another
A digital filter's output y(k) is related to its input x(k) by
convolution with its impulse response h(k).
Cont.
x = [1 2 1];
h = [1 1 1];
y = conv(h,x);
stem(y)
Filters and Transfer Functions
In general, the z-transform Y(z) of a digital filter's output y(n) is
related to the z-transform X(z) of the input by
H(z)
where H(z) is the filter's transfer function. Here, the constants
b(i) and a(i) are the filter coefficients and the order of the filter is
the maximum of n and m.
Cont.
For example let;
1
H ( z)
1 0.9 z 1
step = ones(50);
%input data : step function
b = 1;
% Numerator
a = [1 -0.9]; % Denominator
where the vectors b and a represent the coefficients of a filter in transfer
function form. To apply this filter to your data, use
y = filter(b,a,step);
stem(y)
fvtool(b,a)
%GUI.Dont have to define input (if input is
%step/impulse function)%
Filter Design and
Implementation
Filter design is the process of creating the filter
coefficients to meet specific filtering requirements. Filter
implementation involves choosing and applying a
particular filter structure to those coefficients.
Only after both design and implementation have been
performed can data be filtered.
Filter Coefficients and Filter Names
In general, the z-transform Y(z) of a digital filter's output y(n) is related to
the z-transform X(z) of the input by
Many standard names for filters reflect the number of a and b coefficients
present:
When n = 0 (that is, b is a scalar), the filter is an Infinite Impulse
Response (IIR), all-pole, recursive, or autoregressive (AR) filter.
When m = 0 (that is, a is a scalar), the filter is a Finite Impulse
Response (FIR), all-zero, nonrecursive, or moving-average (MA)
filter.
If both n and m are greater than zero, the filter is an IIR, pole-zero,
recursive, or autoregressive moving-average (ARMA) filter.
IIR Filter Design
The primary advantage of IIR filters over FIR filters is that they
typically meet a given set of specifications with a much lower filter
order than a corresponding FIR filter
Complete Classical IIR Filter Design
Filter Type
Design Function
Bessel (analog only)
[b,a] = besself(n,Wn,options)
[z,p,k] = besself(n,Wn,options)
[A,B,C,D] = besself(n,Wn,options)
Butterworth
[b,a] = butter(n,Wn,options)
[z,p,k] = butter(n,Wn,options)
[A,B,C,D] = butter(n,Wn,options)
Chebyshev Type I
[b,a] = cheby1(n,Rp,Wn,options)
[z,p,k] = cheby1(n,Rp,Wn,options)
[A,B,C,D] = cheby1(n,Rp,Wn,options)
Chebyshev Type II
[b,a] = cheby2(n,Rs,Wn,options)
[z,p,k] = cheby2(n,Rs,Wn,options)
[A,B,C,D] = cheby2(n,Rs,Wn,options)
Elliptic
[b,a] = ellip(n,Rp,Rs,Wn,options)
[z,p,k] = ellip(n,Rp,Rs,Wn,options)
[A,B,C,D] = ellip(n,Rp,Rs,Wn,options)
Cont
Example 1:
For data sampled at 1000 Hz, design a 9th-order highpass
Butterworth IIR filter with cutoff frequency of 300 Hz,
Solution:
9th order IIR filter
Specifies cut off freq., normalised to half sampling
freq. (Nyquist Theorem)
[b,a] = butter(9,300/500,'high');
freqz(b,a,128,1000)
Highpass
filter
Cont.
Example 2:
For data sampled at 1000 Hz, design a 9th-order lowpass
Chebyshev Type I filter with 0.5 dB of ripple in the
passband and a cutoff frequency of 300 Hz, which
corresponds to a normalized value of 0.6:
Solution
[b,a] = cheby1(9,0.5,300/500);
freqz(b,a,512,1000)%The frequency response of the filter
FIR Filter Design
Digital filters with finite-duration impulse response (all-zero, or FIR
filters) have both advantages and disadvantages compared to infiniteduration impulse response (IIR) filters.
FIR filters have the following primary advantages:
They can have exactly linear phase.
They are always stable.
The design methods are generally linear.
They can be realized efficiently in hardware.
The filter startup transients have finite duration.
The primary disadvantage of FIR filters is that they often require a much
higher filter order than IIR filters to achieve a given level of
performance. Correspondingly, the delay of these filters is often much
greater than for an equal performance IIR filter.
Method
Description
Function
s
Windowing
Apply window to truncated inverse Fourier
transform of desired "brick wall" filter
fir1, fir2,
kaiserord
Multiband with
Transition Bands
Equiripple or least squares approach over
sub-bands of the frequency range
firls, remez
, remezord
Constrained
Least Squares
Minimize squared integral error over
entire frequency range subject to
maximum error constraints
fircls,
fircls1
Arbitrary
Response
Arbitrary responses, including nonlinear
phase and complex filters
cremez
Raised Cosine
Lowpass response with smooth,
sinusoidal transition
firrcos
Cont.
Example 1
Design a 48th-order FIR bandpass filter with passband 0.35
0.65:
Solution
b = fir1(48,[0.35 0.65]);
freqz(b,1,512)
Cont.
Example 2:
Design a lowpass filter with the following specifications using
the optimal design method :
rp = 0.01;
rs = 0.1;
fs = 8000;
f = [1500 2000];
a = [1 0];
% Passband ripple
% Stopband ripple
% Sampling frequency
% Cutoff frequencies
% Desired amplitudes
Cont.
Solution
a vector of maximum
deviations or
ripples allowable
for each band.
%[n,fo,ao,w] = remezord(f,a,dev,fs); %
dev=[0.01 0.1]
[n,fo,ao,w]=remezord([1500 2000],[1 0],dev,8000); %
approximate order, normalized frequency band edges, frequency band amplitudes, and
weights that meet input specifications f, a, and dev.%
b=remez(n,fo,ao,w);
%use n, fo, ao and w to design the filter
b which approximately meets the specifications given by remezord input
parameters f, a, and dev.%
freqz(b,1,1024,8000);
title('Lowpass Filter Designed to Specifications');
Filter Implementation
After the filter design process has generated the filter coefficient
vectors, b and a, two functions are available in the Signal Processing
Toolbox for implementing the filter:
filter-:for b and a coefficient input, implements a direct-form II
transposed structure and filters the data. For dfilt input, filter
uses the structure specified with dfilt and filters the data.
Dfilt-:let us specify the filter structure and creates a digital filter
object.
Cont.
Example using filter
t = (0:0.001:1);
%fs=1/0.001Hz
x = sin(2*pi*50*t) + 2*sin(2*pi*120*t); %define input
plot(t,x)
%plot input
[b,a] = butter(9,100/500,'high');
%9th order,high pass filter
%with cutoff freq. 100Hz
c=filter(b,a,x);
figure(2)
plot(t,c)
Cont.
The complete process of designing, implementing, and applying a
filter using a dfilt object is described below:
1.
2.
3.
Generate the filter coefficients using any IIR or FIR filter
design function.
Create the filter object from the filter coefficients and the
specified filter structure using dfilt.
Apply the dfilt filter object to the data, x using filter .
Cont.
Example using dfilt
t = (0:0.001:1);
x = sin(2*pi*50*t) + 2*sin(2*pi*120*t); %define input
plot(t,x)
[b,a] = butter(9,100/500,'high');
%design filter
Hd = dfilt.df2t(b,a);
%Implement direct%form II transposed
c=filter(Hd,x);
returns a discrete-time
figure(2)
filter object, Hd, of type
plot(t,c)
df2t(direct-form II
transposed
Statistical Signal Processing
Correlation and Covariance
The functions xcorr and xcov estimate the cross-correlation and crosscovariance sequences of random processes .
The cross-correlation sequence is a statistical quantity defined as
,
where xn and yn are stationary random processes,
, and E{} is the
expected value operator which measures the similarity between the two
waveforms.The covariance sequence is the mean-removed cross-correlation
sequence
Cont.
or, in terms of the cross-correlation,
Cont.
Example on xcorr
x = [1 1 1 1 1]';
y = x;
xyc = xcorr(x,y)
stem(xyc)
Example on xcov
ww = randn(1000,1); % Generate uniform noise with mean = 1/2.%
[cov_ww,lags] = xcov(ww,10,'coeff');
stem(lags,cov_ww)
Fast Fourier Transform (FFT)
The discrete Fourier transform, or DFT, is the primary
tool of digital signal processing. The foundation of the
Signal Processing Toolbox is the fast Fourier transform
(FFT), a method for computing the DFT with reduced
execution time. Many of the toolbox functions (including
z-domain frequency response, spectrum and cepstrum
analysis, and some filter design and implementation
functions) incorporate the FFT.
Cont.
t = (0:0.001:1);
%0.001 is sampling
x = sin(2*pi*50*t) + 2*sin(2*pi*120*t);
y = fft(x);
%Compute DFT of x
m = abs(y);
%magnitude
f = (0:length(y)/2-1)*1000/length(y); %Frequency vector
plot(f,m(1:1:(length(m)-1)/2))
%before filter
grid
[b,a] = butter(9,100/500,'high');
%design filter
c=filter(b,a,x);
%implement filter
figure(2)
y = fft(c);
%Compute DFT of c(filtered x)
m = abs(y);
% Magnitude
f = (0:length(y)/2-1)*1000/length(y); %Frequency vector
plot(f,m(1:1:(length(m)-1)/2))
%after filter
Grid
FDATool: A Filter Design and
Analysis GUI
The Filter Design and Analysis Tool (FDATool) is a powerful user
interface for designing and analyzing filters. FDATool enables you to
quickly design digital FIR or IIR filters by setting filter performance
specifications, by importing filters from your MATLAB workspace,
or by directly specifying filter coefficients.
FDATool also provides tools for analyzing filters, such as magnitude
and phase response plots and pole-zero plots. You can use FDATool
as a convenient alternative to the command line filter design
functions.
Opening FDATool
To open the Filter Design and Analysis Tool, type
fdatool
Choosing a Response Type
You can choose from several response types:
Choosing a Filter Design Method
You can use the default filter design method for the response type
that you've selected, or you can select a filter design method from the
available FIR and IIR methods listed in the GUI.
To select the Remez algorithm to compute FIR filter coefficients,
select the FIR radio button and choose Equiripple from the list of
methods.
Setting the Filter Design Specifications
The filter design specifications that you can set vary according to
response type and design method. For example, to design a bandpass
filter, you can enter
Filter order
Options
Bandpass Filter Frequency Specifications
Bandpass Filter Magnitude Specifications
Bandpass Filter Frequency Specifications
For this example:
Bandpass Filter Magnitude Specifications
For this example
Computing the Filter Coefficients
Now that you've specified the filter design, click the Design Filter button to
compute the filter coefficients .
References
1. Tood K. Moon, Wynn C. Stirling
Mathematical Methods and Algorithms
for Signal Processing,Prentice
Hall,2000.
2. Samuel D. Stearns, Ruth A. David
Signal Processing Algorithms In
Matlab, Prentice Hall,1996.
3. http://www.mathworks.com.
THANK YOU
