FFT2 Algorithm
FFT2 Algorithm
DISCRETE FOURIER TRANSFORM (DFT) AND FAST FOURIER TRANSFORM (FFT) ALGORITHM
Lectured by Assoc Prof. Dr. Thuong Le-Tien
But if the replicas overlap they will contribute to the right hand side of spectrum
(f) In terms of the time samples x(nT), the original sampled spectrum X
( f ) are given by: and its time-windowed version X L
Setting W(n) = 1
Frequency resolution
These equation can be rewritten to give the minimum number of samples required To achieve a desired freq resolution f. The smaller the desired separation, the Longer the data record
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 6
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
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 9
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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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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
DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien 13
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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And the 2-point and 4ponit 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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Example: Determine the mod-4 and mod-3 reduction of the length-8 signal vector
For n=0, 1, 2,
, N-1
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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:
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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~
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
Where IN is the N-dimensional identity matrix and A is the complex ~ ~ conjugate of A , obtained by conjugating every matrix element of A . For example, for N=4, we can verify easily:
~*
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, we have Xk = X(
) and
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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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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
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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. Each of these is replaced by two (N/4)-point DFTs , and so on.
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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 seperately. 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
0 2n 1 N 1 1 2n 1 n 1 0 2n N 2
DSP lectured by Assoc. Prof. Dr. Thuong Le-Tien
0 n
N /2 1
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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 one-dimensional 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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