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LTI System (Linear Time Invariant) Implementation

The document discusses Fourier analysis techniques. It explains that [1] the Fourier transform represents any signal as a sum of sinusoids with different frequencies, [2] the discrete Fourier transform (DFT) calculates the coefficients for a finite number of samples, and [3] the fast Fourier transform (FFT) computes the DFT more efficiently in O(n log n) time instead of O(n2) time for the DFT. An example is provided to demonstrate calculating the amplitude and phase of a signal using the DFT.

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0% found this document useful (0 votes)
65 views16 pages

LTI System (Linear Time Invariant) Implementation

The document discusses Fourier analysis techniques. It explains that [1] the Fourier transform represents any signal as a sum of sinusoids with different frequencies, [2] the discrete Fourier transform (DFT) calculates the coefficients for a finite number of samples, and [3] the fast Fourier transform (FFT) computes the DFT more efficiently in O(n log n) time instead of O(n2) time for the DFT. An example is provided to demonstrate calculating the amplitude and phase of a signal using the DFT.

Uploaded by

saRIKA
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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LTI system

(Linear Time Invariant)


Implementation
Fourier Series
• F(t) = ½ a0 + sum (ak*cos(2*pi*k*t) + bk*sin(2*pi*k*t))
• Where frequency k =1 to infinity
• Two components here are
• Constant
• Sum of sine and cosine (sinusoids with all frequencies)

• Any signal can be represented in this way


• Problem is to calculate coefficients ak and bk for every frequency
• Fourier Transform
Fourier Transform
• Multiply function with analyzing function

• Here analyzing function is sinusoids

• Here sinusoids are represented as expression of complex coefficient

• One complex coefficient per frequency


Fourier Transform

• Here time is varying from –infinity to + infinity

• In practice we have a discrete signal so Discrete Fourier Transform is


used
Discrete Fourier Transform

• Transform for N samples, where n=0 to N-1

• We are searching coefficient for each frequency k


• K varies from 1 to sampling freq with interval = resolution of the signal
Discrete Fourier Transform
• Unfold the summation

• bn = 2*pi*k*n/N

• Also by Euler’s formula


• e^(j*x) = cos x + j sin x

• X(k) = x0 [ cos(-b0) + j sin(-b0)] + x1[ cos(-b1) + j sin(-b1)] + ..


• X(k) = A(k) + B(k) j … one complex no with real and imaginary part
Discrete Fourier Transform
• X(k) = A(k) + B(k) j
• Maginitude / Amplitude of Sinusoid at freq K is
• sqrt(a(k)^2 + b(k)^2)
• Phase / Shift of Sinusoid at freq K is
• Tan_inverse(b(k) / a(k))

• Ref: https://www.youtube.com/watch?v=mkGsMWi_j4Q
Discrete Fourier Transform Example
• Question Find the Amplitude and Phase using DFT by using following signal
• X0 = 0
• X1= 0.707
• X2= 1
• X3= 0.707
• X4= 0
• X5 = -0.707
• X6 = -1
• X7 = -0.707

• Reading NOTE: This is 1Hz Sine wave with amplitude 1


• Sampling freq = 8Hz
• So, samples in 1 sec = 8
Discrete Fourier Transform Example
• So, we can plot frequencies ,
• Freq resolution = sampling freq / no of samples

• So for k=0 ,
• X(0) = x(0) * e ^0 + x(1) * e^ 0 + …
• X(0) = 0 + 0.707 + .. [sum of all samples]
• X(0)= 0
• ----------
• X(1) = x(0) * e ^ -((j * 2pi*1*0)/8) + x(1) * e ^ -((j * 2pi*1*1)/8)+..
• X(1) = 0+0.707 [ cos(-pi/4) + j * sin(-pi/4)] + …
• X(1) = -4j
Discrete Fourier Transform Example
• So for k=2 -6
• X(2) = 0
• ----------
• k=7
• X(7) = 4j
• -----
• Nyquist limit :: states that we can’t measure frequencies beyond sampling freq /
2
• So, all the values before the Nyquist limit are doubled and all values after it are
removed
• So final DFT are
• x(0)= 0, x(1) = -8j, x(2) = 0, x(3) = 0 .. All values fromx(4) are removed
Discrete Fourier Transform Example
• Amplitude(k) = sqrt(a(k)^2 + b(k)^2)
• ampl(0) = 0, ampl(1) = 8, ampl(2) = 0, ..same 0 till 8
• Avg Amplitude of the Signal = Sum of all amplitudes / no of samples
= (0+8+0+0+0+0+0+0)/8

• Avg Amplitude of the Signal = 8/8 = 1

• At freq =1, Phase(1) = tan_inv( -8/0)


• So we need to find phase by plotting it,
• Phase ( 1) = 3*pi/2
Discrete Fourier Transform

• Transform for N samples, where n=0 to N-1

• We are searching coefficient for each frequency k


Discrete Fourier Transform
• Discrete in time as well as frequency domain

• In DFT N*N complex number multiplications so O(n^2)

• It can be represented in matrix multiplication format

• [coefficient matrix](size=n*n) * [ input signal vector x0-xn](Size=1*n)

• Every coefficient is e ^ (-j*2*pi*k*n*/N)


Fast Fourier Transform
• Very fast O(n log n )  streaming of signal becomes easy
• Radix 2 Algorithm

• Length of Input N = 2^v


• Split the seq into two parts, even sample and odd sample
• Once two signals are there then calculate the DFT

• How to do it?
• Reverse the bits of index in bit reversal mode
FFT Circuit
Signal Split in Original Bit Revesed
Index odd Index Index
Even 000 000
0 0 0
001 100
1 2 4
010 010
2 4 2
011 110
3 6 6
100 001
4 1 1
5 3 5 101 101
6 5 3 110 011
7 7 7 111 111
FFT Circuit

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