American International University

Digital Signal Processing Lab

Experiement#9

Experiment no. 4( Final Term): Fast Fourier Transforms (FFT)

Objective:

The objective of this experiment is to become familiar with Fast Fourier Transform (FFT)


Introduction

The Fast Fourier transform (FFT) is an algorithm for computing DFT, before which the
DFT required excessive amount of computation time, particularly when high number of
samples (N) was required. The FFT forces one further assumption, that N is an integer
multiple of 2. This allows certain symmetries to occur reducing the number of
calculations (especially multiplications) which have to be done

Matlab's FFT function is an effective tool for computing the discrete Fourier transform of
a signal. The following code examples will help you to understand the details of using
the FFT function.

Example 1:

The typical syntax for computing the FFT of a signal is FFT(x,N) where x is the signal,
x[n], you wish to transform, and N is the number of points in the FFT. N must be at least
as large as the number of samples in x[n]. To demonstrate the effect of changing the
value of N, synthesize a cosine with 30 samples at 10 samples per period.

n =[0:29];
x =cos(2*pi*n/10);

Define 3 different values for N. Then take the transform of x[n] for each of the 3 values
that were defined . The abs function absolute of the magnitude of the transform, as we
are not concerned with distinguishing between real and imaginary components.

N1 =64;
N2 =128;
N3 =256;
X1 =abs(fft(x,N1));
X2 =abs(fft(x,N2));
X3 =abs(fft(x,N3));

The frequency scale begins at 0 and extends to N-1 for an N-point FFT. We then
normalize the scale so that it extends from 0 to 1-1/N
.
F1 =[0 : N1 - 1]/N1;
F2 =[0 : N2 - 1]/N2;
F3 =[0 : N3 - 1]/N3;
Plot each of the transforms one above the other.
subplot(3,1,1)
plot(F1,X1,'-x'),title('N =64'),axis([0 1 0 20])
subplot(3,1,2)
plot(F2,X2,'-x'),title('N =128'),axis([0 1 0 20])
subplot(3,1,3)
plot(F3,X3,'-x'),title('N =256'),axis([0 1 0 20])


0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
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N = 64
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
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N = 128
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
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N = 256





Upon examining the plot (shown in in Figure ) one can see that each of the transforms
adheres to the same shape, differing only in the number of samples used to approximate
that shape. What happens if N is the same as the number of samples in x[n]? To find out,
set N1 =30. What does the resulting plot look like? Why does it look like this?
Example 2:

In the the last example the length of x[n] was limited to 3 periods in length. Now, let's
choose a large value for N (for a transform with many points), and vary the number of
repetitions of the fundamental period.

n =[0:29];
x1 =cos(2*pi*n/10); % 3 periods
x2 =[x1 x1]; % 6 periods
x3 =[x1 x1 x1]; % 9 periods
N =2048;
X1 =abs(fft(x1,N));
X2 =abs(fft(x2,N));
X3 =abs(fft(x3,N));
F =[0:N-1]/N;
subplot(3,1,1)
plot(F,X1),title('3 periods'),axis([0 1 0 50])
subplot(3,1,2)
plot(F,X2),title('6 periods'),axis([0 1 0 50])
subplot(3,1,3)
plot(F,X3),title('9 periods'),axis([0 1 0 50])
The previous code will produce three plots as shown in the following figure.
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
0
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3 periods
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
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6 periods
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
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9 periods



The first plot, the transform of 3 periods of a cosine, looks like the magnitude of 2 sincs
with the center of the first sinc at 0.1fs and the second at 0.9fs. The second plot also has a
sinc-like appearance, but its frequency is higher and it has a larger magnitude at 0.1fs and
0.9fs. Similarly, the third plot has a larger sinc frequency and magnitude.

As x[n] is extended to an large number of periods, the sincs will begin to look more and
more like impulses. But we know that a sinusoid transformed to an impulse, why do we
have sincs in the frequency domain? When the FFT is computed with an N larger than the
number of samples in x[n], it fills in the samples after x[n] with zeros. Example 2 had an
x[n] that was 30 samples long, but the FFT had an N =2048. When Matlab computes the
FFT, it automatically fills the spaces from n =30 to n =2047 with zeros. This is like
taking a sinusoid and multiplying it with a rectangular box of length 30. A multiplication
of a box and a sinusoid in the time domain should result in the convolution of a sinc with
impulses in the frequency domain. Furthermore, increasing the width of the box in the
time domain should increase the frequency of the sinc in the frequency domain. The
previous Matlab experiment supports this conclusion.

Example 4: Spectrum Analysis with the FFT and Matlab
The FFT does not directly give you the spectrum of a signal. As we have seen with the
last two experiments, the FFT can vary dramatically depending on the number of points
(N) of the FFT, and the number of periods of the signal that are represented. There is
another problem as well. The FFT contains information between 0 and fs, however, we
know that the sampling frequency 5 must be at least twice the highest frequency
component. Therefore, the signal's spectrum should be entirely below fs, the Nyquist
frequency. Recall also that a real signal should have a transform magnitude that is
symmetrical for positive and negative frequencies. So instead of having a spectrum that
goes from 0 to fs, it would be more appropriate to show the spectrum from fs/2 to fs/2
This can be accomplished by using Matlab's shift function as the following code
demonstrates.
n =[0:149];
x1 =cos(2*pi*n/10);
N =2048;
X =abs(fft(x1,N));
X =fftshift(X);
F =[-N/2:N/2-1]/N;
plot(F,X),
xlabel('frequency / f s')
0.5 0.4 0.3 0.2 0.1 0 0.1 0.2 0.3 0.4 0.5
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frequency / f s