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I have adapted the code in Comparing FFT of Function to Analytical FT Solution in Matlab for this question. I am trying to do FFTs and comparing the result with analytical expressions in the Wikipedia tables.
My code is:
a = 1.223;
fs = 1e5; %sampling frequency
dt = 1/fs;
t = 0:dt:30-dt; %time vector
L = length(t); % no. sample points
t = t - 0.5*max(t); %center around t=0
y = ; % original function in time
Y = dt*fftshift(abs(fft(y))); %numerical soln
freq = (-L/2:L/2-1)*fs/L; %freq vector
w = 2*pi*freq; % angular freq
F = ; %analytical solution
figure; subplot(1,2,1); hold on
plot(w,real(Y),'.')
plot(w,real(F),'-')
xlabel('Frequency, w')
title('real')
legend('numerical','analytic')
xlim([-5,5])
subplot(1,2,2); hold on;
plot(w,imag(Y),'.')
plot(w,imag(F),'-')
xlabel('Frequency, w')
title('imag')
legend('numerical','analytic')
xlim([-5,5])
If I study the Gaussian function and let
y = exp(-a*t.^2); % original function in time
F = exp(-w.^2/(4*a))*sqrt(pi/a); %analytical solution
in the above code, looks like there is good agreement when the real and imaginary parts of the function are plotted:
But if I study a decaying exponential multiplied with a Heaviside function:
H = #(x)1*(x>0); % Heaviside function
y = exp(-a*t).*H(t);
F = 1./(a+1j*w); %analytical solution
then
Why is there a discrepancy? I suspect it's related to the line Y = but I'm not sure why or how.
Edit: I changed the ifftshift to fftshift in Y = dt*fftshift(abs(fft(y)));. Then I also removed the abs. The second graph now looks like:
What is the mathematical reason behind the 'mirrored' graph and how can I remove it?
The plots at the bottom of the question are not mirrored. If you plot those using lines instead of dots you'll see the numeric results have very high frequencies. The absolute component matches, but the phase doesn't. When this happens, it's almost certainly a case of a shift in the time domain.
And indeed, you define the time domain function with the origin in the middle. The FFT expects the origin to be at the first (leftmost) sample. This is what ifftshift is for:
Y = dt*fftshift(fft(ifftshift(y)));
ifftshift moves the origin to the first sample, in preparation for the fft call, and fftshift moves the origin of the result to the middle, for display.
Edit
Your t does not have a 0:
>> t(L/2+(-1:2))
ans =
-1.5000e-05 -5.0000e-06 5.0000e-06 1.5000e-05
The sample at t(floor(L/2)+1) needs to be 0. That is the sample that ifftshift moves to the leftmost sample. (I use floor there in case L is odd in size, not the case here.)
To generate a correct t do as follows:
fs = 1e5; % sampling frequency
L = 30 * fs;
t = -floor(L/2):floor((L-1)/2);
t = t / fs;
I first generate an integer t axis of the right length, with 0 at the correct location (t(floor(L/2)+1)==0). Then I convert that to seconds by dividing by the sampling frequency.
With this t, the Y as I suggest above, and the rest of your code as-is, I see this for the Gaussian example:
>> max(abs(F-Y))
ans = 4.5254e-16
For the other function I see larger differences, in the order of 6e-6. This is due to the inability to sample the Heaviside function. You need t=0 in your sampled function, but H doesn't have a value at 0. I noticed that the real component has an offset of similar magnitude, which is caused by the sample at t=0.
Typically, the sampled Heaviside function is set to 0.5 for t=0. If I do that, the offset is removed completely, and max difference for the real component is reduced by 3 orders of magnitude (largest errors happen for values very close to 0, where I see a zig-zag pattern). For the imaginary component, the max error is reduced to 3e-6, still quite large, and is maximal at high frequencies. I attribute these errors to the difference between the ideal and sampled Heaviside functions.
You should probably limit yourself to band-limited functions (or nearly-band-limited ones such as the Gaussian). You might want to try to replace the Heaviside function with an error function (integral of Gaussian) with a small sigma (sigma = 0.8 * fs is the smallest sigma I would consider for proper sampling). Its Fourier transform is known.
From an lsqcurve fit function I am trying to calculate the horizontal asymptote of a sigmoidal curve over the data range. The code I am using to do this and convert the output to double precision works and is as follows:
time = [0,0.166666666666667,0.333333333333333,0.500000000000000,0.666666666666667,0.833333333333333,1,1.16666666666667,1.33333333333333,1.50000000000000,1.66666666666667,1.83333333333333,2,2.16663888888889,2.33330555555556,2.49997222222222,2.66663888888889,2.83330555555556,2.99997222222222,3.16663888888889,3.33330555555556,3.49997222222222,3.66663888888889,3.83330555555556,3.99997222222222,4.16663888888889,4.33330555555556,4.49997222222222,4.66663888888889,4.83330555555556,4.99997222222222,5.16663888888889,5.33327777777778,5.49997222222222,5.66661111111111,5.83327777777778,5.99994444444444,6.16661111111111,6.33327777777778,6.49994444444444,6.66661111111111,6.83327777777778,6.99994444444444,7.16661111111111,7.33327777777778,7.49994444444444,7.66661111111111,7.83327777777778,7.99994444444444,8.16661111111111,8.33327777777778,8.49994444444445,8.66661111111111,8.83325000000000,8.99991666666667,9.16658333333333,9.33325000000000,9.49991666666667,9.66658333333333,9.83325000000000,9.99991666666667,10.1665833333333,10.3332500000000,10.4999166666667,10.6665833333333,10.8332500000000,10.9999166666667,11.1665833333333,11.3332500000000,11.4999166666667,11.6665833333333,11.8332222222222,11.9998888888889,12.1665555555556,12.3332222222222,12.4998888888889,12.6665555555556,12.8332222222222,12.9998888888889,13.1665555555556,13.3332222222222,13.4998888888889,13.6665555555556,13.8332222222222,13.9998888888889,14.1665555555556,14.3332222222222,14.4998888888889,14.6665555555556,14.8332222222222,14.9998611111111,15.1665277777778,15.3331944444444,15.4998611111111];
Abs=[0.0756333346168200,,0.0822666635115941,0.101066668828328,0.125066662828128,0.147433335582415,0.169966672857602,0.192199997603893,0.211166655023893,0.231433339416981,0.256133330365022,0.279866663118203,0.295566670596600,0.309466672440370,0.323400000731150,0.335233343144258,0.348299992581209,0.359700001776218,0.371433332562447,0.383333327869574,0.394433322052161,0.404166663686434,0.413666687905788,0.423499998946985,0.432733344535033,0.443266667425633,0.453899989525477,0.463699989020824,0.475066669285297,0.485300009449323,0.495999989410241,0.504966656366984,0.514033327500025,0.523766654233138,0.532566664119562,0.541033337513606,0.550333333512147,0.559733323752880,0.567733330031236,0.575466672579447,0.583399988710880,0.591433353722096,0.597833327949047,0.604699979225794,0.610233349104722,0.616733354826768,0.622800009946028,0.628666676580906,0.634199996789297,0.638966679573059,0.644799992442131,0.650733329355717,0.654566655556361,0.661333315074444,0.666366661588351,0.671233326196671,0.675699991484483,0.679433315992355,0.682733337084452,0.686333330969016,0.689166650176048,0.693133344252904,0.697366672257582,0.699333322544893,0.702233329415321,0.705400002499422,0.707699996729692,0.709999985992909,0.711533337831497,0.712800003588200,0.713633326192697,0.714666672050953,0.714533333977064,0.713999989132086,0.714266665279865,0.714066684246063,0.714266670246919,0.713733327885469,0.713066659867764,0.712200003365676,0.711600005626679,0.710400002698104,0.710399977862835,0.709799999992053,0.708366667230924,0.706933336953322,0.706766667465369,0.706199998656909,0.704266687234243,0.703333348035812,0.702399998903275,0.701133318245411,0.700166669984659,0.698799995084604,0.697700018684069];
u=linspace(0,15.49986111,500);
v=pchip(time,Abs,u);
p0 = [0.075633335,0.05,0.01]; %First value = Abs value at t=0
options=optimset('MaxFunEvals',10000,'MaxIter',5000,'Display','iter');
[p,error,residual]=lsqcurvefit(#gompertz,p0,time,Abs,[],[],options)
syms x t
f=p(1)*exp(-exp(((p(2)*exp(1))/p(1))*(p(3)-t)+1));
f1=diff(f,t); %Find first derivative of equation W wrt t
f2=diff(f,t,2); %Find second derivative wrt t and to allow inflection point to be defined
crit_pts = solve(f1); %Find local maxima and minima
crit_pts=double(crit_pts);
flimit = limit(f,inf); %Find find horizontal asymptote of function f by taking limit of f as x approaches infinity
function_horizontal_asymptote=double(limit);
The 'gompertz' function called is coded as follows:
function y = gompertz(p,t)
y=p(1)*exp(-exp(((p(2)*exp(1))/p(1))*(p(3)-t)+1));
According to the Matlab documentation I have read
flimit = limit(f,inf);
should yield the correct answer but and appear towards the end of the time course, however the result I get is (shown by the yellow marker):
Am I calculating this incorrectly?
Thank you for your time,
Laura
I'm new to programming in Matlab. I'm trying to figure out how to calculate the following function:
I know my code is off, I just wanted to start with some form of the function. I have attempted to write out the sum of the function in the program below.
function [g] = square_wave(n)
g = symsum(((sin((2k-1)*t))/(2k-1)), 1,n);
end
Any help would be much appreciated.
Update:
My code as of now:
function [yout] = square_wave(n)
syms n;
f = n^4;
df = diff(f);
syms t k;
f = 1; %//Define frequency here
funcSum = (sin(2*pi*(2*k - 1)*f*t) / (2*k - 1));
funcOut = symsum(func, v, start, finish);
xsquare = (4/pi) * symsum(funcSum, k, 1, Inf);
tVector = 0 : 0.01 : 4*pi; %// Choose a step size of 0.01
yout = subs(xsquare, t, tVector);
end
Note: This answer was partly inspired by a previous post I wrote here: How to have square wave in Matlab symbolic equation - However, it isn't quite the same, which is why I'm providing an answer here.
Alright, so it looks like you got the first bit of the question right. However, when you're multiplying things together, you need to use the * operator... and so 2k - 1 should be 2*k - 1. Ignoring this, you are symsuming correctly given that square wave equation. The input into this function is only one parameter only - n. What you see in the above equation is a Fourier Series representation of a square wave. A bastardized version of this theory is that you can represent a periodic function as an infinite summation of sinusoidal functions with each function weighted by a certain amount. What you see in the equation is in fact the Fourier Series of a square wave.
n controls the total number of sinusoids to add into the equation. The more sinusoids you have, the more the function is going to look like a square wave. In the question, they want you to play around with the value of n. If n becomes very large, it should start approaching what looks like to be a square wave.
The symsum will represent this Fourier Series as a function with respect to t. What you need to do now is you need to substitute values of t into this expression to get the output amplitude for each value t. They define that for you already where it's a vector from 0 to 4*pi with 1001 points in between.
Define this vector, then you'll need to use subs to substitute the time values into the symsum expression and when you're done, cast them back to double so that you actually get a numeric vector.
As such, your function should simply be this:
function [g] = square_wave(n)
syms t k; %// Define t and k
f = sin((2*k-1)*t)/(2*k-1); %// Define function
F = symsum(f, k, 1, n); %// Define Fourier Series
tVector = linspace(0, 4*pi, 1001); %// Define time points
g = double(subs(F, t, tVector)); %// Get numeric output
end
The first line defines t and k to be symbolic because t and k are symbolic in the expression. Next, I'll define f to be the term inside the summation with respect to t and k. The line after that defines the actual sum itself. We use f and sum with respect to k as that is what the summation calls for and we sum from 1 up to n. Last but not least, we define a time vector from 0 to 4*pi with 1001 points in between and we use subs to substitute the value of t in the Fourier Series with all values in this vector. The result should be a 1001 vector which I then cast to double to get a numerical result and we get your desired output.
To show you that this works, we can try this with n = 20. Do this in the command prompt now:
>> g = square_wave(20);
>> t = linspace(0, 4*pi, 1001);
>> plot(t, g);
We get:
Therefore, if you make n go higher... so 200 as they suggest, you'll see that the wave will eventually look like what you expect from a square wave.
If you don't have the Symbolic Math Toolbox, which symsum, syms and subs relies on, we can do it completely numerically. What you'll have to do is define a meshgrid of points for pairs of t and n, substitute each pair into the sequence equation for the Fourier Series and sum up all of the results.
As such, you'd do something like this:
function [g] = square_wave(n)
tVector = linspace(0, 4*pi, 1001); %// Define time points
[t,k] = meshgrid(tVector, 1:n); %// Define meshgrid
f = sin((2*k-1).*t)./(2*k-1); %// Define Fourier Series
g = sum(f, 1); %// Sum up for each time point
end
The first line of code defines our time points from 0 to 4*pi. The next line of code defines a meshgrid of points. How this works is that for t, each column defines a unique time point, so the first column is 200 zeroes, up to the last column which is a column of 200 4*pi values. Similarly for k, each row denotes a unique n value so the first row is 1001 1s, followed by 1001 2s, up to 1001 1s. The implications with this is now each column of t and k denotes the right (t,n) pairs to compute the output of the Fourier series for each time that is unique to that column.
As such, you'd simply use the sequence equation and do element-wise multiplication and division, then sum along each individual column to finally get the square wave output. With the above code, you will get the same result as above, and it'll be much faster than symsum because we're doing it numerically now and not doing it symbolically which has a lot more computational overhead.
Here's what we get when n = 200:
This code with n=200 ran in milliseconds whereas the symsum equivalent took almost 2 minutes on my machine - Mac OS X 10.10.3 Yosemite, 16 GB RAM, Intel Core i7 2.3 GHz.
Is there an easy way to calculate the frequency response of the following function?
I tried using heaviside function but with no luck.
Basically I want to write a function to return the frequency response based on input N1 and N2 and also the number of points (lets say x) between 0 and pi
The output would be a vector which returns x values for the frequency response for corresponding frequencies => 0:pi/x:pi
Assuming that N1 + N2 < num_points, where num_points is the length of the sequence, you can simply write the function like so:
function [gr] = rosenburg(N1, N2, num_points)
gr = zeros(num_points,1);
range1 = 0:N1;
range2 = N1+1:N1+N2;
gr(range1+1) = 0.5*(1 - cos(pi*range1/N1));
gr(range2+1) = cos(pi*(range2-N1) / (2*N2));
end
The function prototype, rosenburg takes in N1, N2 and the total number of points you want this function to take in, num_points. How this code works is that we first allocate an array that is all zeroes of size num_points. We then compute two linear ranges: One from 0 <= n <= N1 and the other from N1 < n <= N2. Note that the second range starts by offsetting N1 by 1 because we have already computed the value at n = N1. Once we compute these ranges, we simply apply the right relationship in the right ranges. Note that when I'm assigning the relationships to the correct intervals in the array, I need to offset by 1 because MATLAB begins indexing arrays at index 1. The rest of the values are zero due to the initialization at the beginning of the function.
Now, if you want to find the frequency response of this signal, just use fft which is the Fast Fourier Transform. It's the classic method to find the frequency domain version of a discrete input signal on a numerical basis. As such, once you create your signal using the rosenburg function, then throw this into the FFT function. How you call it is like so:
X = fft(gr);
This computes the N point FFT, where N is the length of the signal gr. Alternatively, you can provide the number of points you want to compute the FFT for. Specifically:
X = fft(gr, N);
Basically, the higher N is, the finer or granular the frequency components will be. Note that the frequency axis is normalized between 0 to 2*pi, and so the higher N is, the finer resolution you will have between neighbouring points on the axis. Specifically, each point on this axis has the following frequency:
w = i*(2*pi)/x;
i would be the index on the x-axis (0, 1, 2, ..., num_points-1) and x would be the total number of points for the FFT. Normally, people show the spectrum between -pi <= w <= pi, and so some people apply fftshift to shift the spectrum so that the DC component is located at the centre of the spectrum, which is how we naturally perceive the spectrum to be.
When you say "frequency response", I believe you are referring to the magnitude, and so use abs to calculate the complex magnitude of each value, as the fft is generally complex valued. Therefore, assuming that you wish to compute the FFT to be as many points as the length of your signal, and let's say we choose N1 = 4, N2 = 8 and we want 64 points, and we want to plot the spectrum. Simply do this:
gr = rosenburg(4, 8, 64);
X = fft(gr);
Xshift = fftshift(X);
plot(linspace(-pi,pi,64), abs(Xshift));
grid;
The above code will shift the spectrum, then plot its magnitude between -pi to pi. This is what I get:
As an illustration, this is what the spectrum looks like before we apply fftshift:
Here's the code to generate the above figure:
plot(linspace(0,2*pi,64), abs(X));
grid;
You can see that the spectra is symmetric. Right at the frequency pi, you can see that it is mirror reflected, which makes sense as the range from pi to 2*pi, precisely maps to -pi to 0. Because the signal is real, the spectrum is symmetric. In fact, we can call this signal Hermitian symmetric. Obviously, the frequency components are a bit sparsely spaced. It may be better to increase the total number of points to something like 256. This is what I get when I change the number of points to 256:
Pretty smooth! Now, if you want to extract the frequency components from 0 to pi, you need to extract half of the frequency decomposition that is stored in X. Therefore, you would simply do:
f = X(1:numel(X)/2);
numel determines how many elements are in an array or matrix. However, remember that each frequency point was defined as:
w = i*(2*pi)/x
You specifically want:
w = i*pi/x
As such, you'll need to compute the FFT at twice the size of your signal first, then extract half of the spectra in the same way. For example, for 64 points:
gr = rosenburg(4, 8, 64);
X = fft(gr, 128);
f = X(1:numel(X)/2);
This should hopefully get you started. Good luck!
I have a weird problem with the discrete fft. I know that the Fourier Transform of a Gauss function exp(-x^2/2) is again the same Gauss function exp(-k^2/2). I tried to test that with some simple code in MatLab and FFTW but I get strange results.
First, the imaginary part of the result is not zero (in MatLab) as it should be.
Second, the absolute value of the real part is a Gauss curve but without the absolute value half of the modes have a negative coefficient. More precisely, every second mode has a coefficient that is the negative of that what it should be.
Third, the peak of the resulting Gauss curve (after taking the absolute value of the real part) is not at one but much higher. Its height is proportional to the number of points on the x-axis. However, the proportionality factor is not 1 but nearly 1/20.
Could anyone explain me what I am doing wrong?
Here is the MatLab code that I used:
function [nooutput,M] = fourier_test
Nx = 512; % number of points in x direction
Lx = 50; % width of the window containing the Gauss curve
x = linspace(-Lx/2,Lx/2,Nx); % creating an equidistant grid on the x-axis
input_1d = exp(-x.^2/2); % Gauss function as an input
input_1d_hat = fft(input_1d); % computing the discrete FFT
input_1d_hat = fftshift(input_1d_hat); % ordering the modes such that the peak is centred
plot(real(input_1d_hat), '-')
hold on
plot(imag(input_1d_hat), 'r-')
The answer is basically what Paul R suggests in his second comment, you introduce a phase shift (linearly dependent on the frequency) because the center of the Gaussian described by input_1d_hat is effectively at k>0, where k+1 is the index into input_1d_hat. Instead if you center your data (such that input_1d_hat(1) corresponds to the center) as follows you get a phase-corrected Gaussian in the frequency domain:
Nx = 512; % number of points in x direction
Lx = 50; % width of the window containing the Gauss curve
x = linspace(-Lx/2,Lx/2,Nx); % creating an equidistant grid on the x-axis
%%%%%%%%%%%%%%%%
x=fftshift(x); % <-- center
%%%%%%%%%%%%%%%%
input_1d = exp(-x.^2/2); % Gauss function as an input
input_1d_hat = fft(input_1d); % computing the discrete FFT
input_1d_hat = fftshift(input_1d_hat); % ordering the modes such that the peak is centered
plot(real(input_1d_hat), '-')
hold on
plot(imag(input_1d_hat), 'r-')
From the definition of the DFT, if the Gaussian is not centered such that maximum occurs at k=0, you will see a phase twist. The effect off fftshift is to perform a circular shift or swapping of left and right sides of the dataset, which is equivalent to shifting the center of the peak to k=0.
As for the amplitude scaling, that is an issue with the definition of the DFT implemented in Matlab. From the documentation for the FFT:
For length N input vector x, the DFT is a length N vector X,
with elements
N
X(k) = sum x(n)*exp(-j*2*pi*(k-1)*(n-1)/N), 1 <= k <= N.
n=1
The inverse DFT (computed by IFFT) is given by
N
x(n) = (1/N) sum X(k)*exp( j*2*pi*(k-1)*(n-1)/N), 1 <= n <= N.
k=1
Note that in the forward step the summation is not normalized by N. Therefore if you increase the number of points Nx in the summation while keeping the width Lx of the Gaussian function constant you will increase X(k) proportionately.
As for signal leaking into the imaginary frequency dimension, that is due to the discrete form of the DFT, which results in truncation and other effects, as noted again by Paul R. If you reduce Lx while keeping Nx constant, you should see a reduction in the amount of signal in the imaginary dimension relative to the real dimension (compare the spectra while keeping peak intensities in the real dimension equal).
You'll find additional answers to similar questions here and here.