DIGITAL SIGNAL PROCESSING

FOURIER TRANSFORM (FT) AND

FAST FOURIER TRANSFORM (FFT)
ALGORITHMS

Lectured by Assoc Prof. Dr. Thuong Le-Tien

September 2011

1. Frequency resolution and windowing

Spectrum of sampled analog signal

But if the replicas overlap they will contribute to the right hand side of spectrum

In terms of the time samples x(nT), the original sampled spectrum X ( f )

and its time-windowed version X L ( f ) are given by:

DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

Rectangular window of length L

Then define the windowed signal

The DTFT of windowed signal is

Where W() is the DTFT of the rectangular window w(n)

DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

Setting W(n) = 1

Magnitude spectrum
of rectangular window

Rectangular window width

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DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

Frequency resolution
To achieve a desired frequency resolution f.
The smaller the desired separation, the
Longer the data record
The Hamming window

At its center, n=(L-1)/2, the value of w(n) is 0.54+0.46 = 1, and at its endpoint,
n=0 and n=L-1, its value is 0.54-0.46 = 0.08
For any type of window, the effective of the
mainlobe is inversely proportional to L
c is a constant and always c=>1
DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

Hamming window in the time and frequency domain

The minimum resolvable frequency difference

DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

Example:
A signal consisting of four sinusoids of frequencies of 1, 1.5, 2.5,
and 2.75 kHz, is sampled at a rate of 10 kHz. What is the minimum
number of samples that should be collected for the frequency
spectrum to exhibit four distinct peaks at these frequencies?
How many samples should be collected if they are going to be
preprocessed by a Hamming window and then Fourier transformed?

Solution:
The smallest frequency separation that must be resolved by the DFT is
f = 2.75-2.5=0.25 kHz, for rectangular window:

Because the mainlobe width of the Hamming window is twice as

wide as that of the rectangular window, it follows that twice as
many samples must be collected, that is L=80 then c can be
calculated to be c=2
DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

Example:
A 10ms portion of a signal is sampled at a rate of 10kHz. It is known
that the signal consists of two sinusoids of frequencies f1=1kHz and
f2=2khz. It is also known that the signal contains a third component
of frequency f3 that lies somewhere between f1 and f2.
a. How close to f1 could f3 be in order for the spectrum of the collected
samples to exhibit three distinct peak? How close to f2 could f3 be?
b.What are the answers if the collected samples are windowed by a
Hamming window?

Solution:
The total number of samples collected is L= fsTL =10x10=100.
The frequency resolution of the rectangular window is
f = fs/L = 10/100 = 0.1kHz
Thus the closest f3 to f1 and f2 will be
f3 = f1 + f = 1.1kHz and f3 = f2 - f = 1.9kHz
In the hamming case, the minimum resolvable frequency separation
doubles, that is,
f = cfs/L = 2.10/100 = 0.2kHz which give f3 = 1.2kHz or f3 = 1.8kHz
DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

2. DTFT computation
2.1. DTFT at a single frequency
DTFT of length-L signal

Rectangular and hamming windows with L=40 and 100

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Equivalent Nyquist Interval

2.2. DFT over frequency range: Compute DFT over

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2.3. DFT
The N points DFT of a L-length signal defined the DFT frequency as follows,

The only difference between DFT and DTFT is that the former has its N
frequencies distributed evenly over the full Nyquist interval [0, 2) whereas
the later has them distributed over any desired subinterval.

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N-point DTFTs over [0,2) and over subinterval [a, b), for N=10

Evaluation of z-transform

The periodicity of X() with a period of 2 or

DFT X(k)=X(k) in the index k with period N
Nth roots of unity for N=8
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2.4. Zeros padding

Note that evaluation at the N frequencies DFT are the same for
the cases of padding D zeros at front or delay D samples

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The DTFT and DFT

2.5. The matrix form of DFT

Denoted
(matrix form of DFT)

Where the matrix components

defined by twiddle factors
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The twiddle factor defined by

For example: L=N and N=2, 4, 8

The corresponding 2-point and 4-point DFT matrices are:

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And the 2-point and 4-point DFT of a length 2 and length 4 signals will be

Thus, the 2-point DFT is formed by taking the sum and difference of the two time
Samples. It will be a convenience starting point for the merging in FFT by hand.

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Twiddle factor look up tables for N=2, 4, 8

5. Modulo N reduction
Example L=4N

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Example: Determine the mod-4 and mod-3 reduction of the length-8 signal vector
For N=4 and N=3

For n=0, 1, 2,

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, N-1

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Periodic extension interpretation of mod-N reduction of a signal

The connection of the mod-N reduction to the DFT is the theorem that the
Length-N wrapped signal x~ has the same N-point DFT as the original
Unwrapped signal x, that is:

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The DFT matrices A and A~ have the same definition, except they
differ in their dimensions, which are NxL and NxN, respectively.
We can write the DFT of x~ in the compact matrix form:

In general A is partitioned in the form:

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N-point DFTs of the full and wrapped signal are equivalent

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Example: Compute the 4-point DFT of the length-8 signal in two way:
(a) Working with the full unwrapped vector x and
(b) Computing the DFT of its mod-4 reduction
Solution: The corresponding DFT is

The same DFT can be computed by the DFT matrix

x~ acted on the wrapped signal x~

The two methods are the same

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6. Inverse DFT
The problem for inverse DFT is the length L of signal greater than N-point DFT,
i.e. the matrix A is not invertible
The inverse DFT defined by

~*
A

Where IN is the N-dimensional identity matrix and

is the complex
~
~
A
conjugate of A , obtained by conjugating every matrix element of
.
For example, for N=4, we can verify easily:

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Similar for FFT

Example for an inverse 4-point DFT

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In term of the DFT frequencies k , we have Xk = X(k ) and

Therefore the alternative form of IDFT

DFT and IDFT

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7. Sampling of periodic signals and DFT

X~ is periodic in n with period N

Discrete Fourier series (DFS)

Sampling rate is a multiple of the fundamental frequency of signal

Taking the Nyquist interval to be the right-sided one [0, fs], we note that
harmonics within that interval are none other than the N DFT frequencies

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Given an integer m, we determine its quotient and reminder of the division

And therefore the corresponding harmonic will be

* Which shows that fm will be aliased with fk. Therefore, if the
signal x(t) is sampled, it will give rise to the samples

Defining the aliased Fourier series amplitudes

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If the sampled signal x(nT) be reconstructed by an ideal reconstructor,

the aliased analog waveform is

Example: determine the aliased signal xal(t) resulting by sampling a square

Wave of frequency f1=1 Hz. For a sampling rate of fs = 4Hz, consider one period
Consisting of N=4 samples and perform its 4-point DFT

The Fourier coefficients:

Corresponding to the harmonic
Where f3 = 3 was replaced by its negative version f3-fs = 3-4 = -1. It follows that
the aliased signal will be

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Similarly, for N=8 corresponding to fs=8 Hz, we perform the 8-point DFT of one
period of the square wave, and divide by 8 to get the aliased amplitudes

These amplitudes corresponding to the frequencies fk = k f1

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8. Fast Fourier Transform FFT

Is a fast implementation of DFT. It is based on a divide and conquer
approach in which the DFT computation is divided into smaller, simpler,
problems and the final DFT is rebuilt from the simpler DFTs.
It is required the initial dimension of N to be power of two

The problem of computing the N-point

DFT is replaced by the simpler problems
of computing two (N/2)-point DFT.

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The summation index n ranges over both even and odd values
in the range [0,N-1]. By grouping the even-indexed and
odd-indexed terms, we get

To determine the proper range of summation over n, we consider the two

Terms separately. For even-indexed terms, the index 2n must be within the
range [0,N-1]. But, because N is even (a power of two), the upper limit
N-1 will be odd. Therefore, the highest even index will be N-2,

0 2n N 2 0 n N / 2 1
Similarly, for the odd-indexed terms, we must have

0 2n 1 N 1

1 2n 1 n 1 0 2n N 2 0 n N / 2 1
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DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien

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The butterfly merging builds

upper and lower halves of
length-N DFT

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and N=8

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The typical algorithm consists of three conceptual parts:

1. Shuffling the N-dimensional input into N of 1-D signals
2. Performing N one-point DFTs
3. Merging the N one-point DFTs into one N-point DFT

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Example: Using FFT algorithm, compute the 4-point

wrapped signal (5, 0, -3, 4)
Solution:
The DFT merging stage merges the two 2-DFTs
into the final 4-DFT

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Example: Using FFT algorithm, compute 8-point DFT of the 8 point signal

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Example: 8-point Inverse FFT

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