Square wave function for Matlab - matlab

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.

Related

How exactly works this simple calculus of a ML gradient descent cost function using Octave\MatLab?

I am following a machine learning course on Coursera and I am doing the following exercise using Octave (MatLab should be the same).
The exercise is related to the calculation of the cost function for a gradient descent algoritm.
In the course slide I have that this is the cost function that I have to implement using Octave:
This is the formula from the course slide:
So J is a function of some THETA variables represented by the THETA matrix (in the previous second equation).
This is the correct MatLab\Octave implementation for the J(THETA) computation:
function J = computeCost(X, y, theta)
%COMPUTECOST Compute cost for linear regression
% J = COMPUTECOST(X, y, theta) computes the cost of using theta as the
% parameter for linear regression to fit the data points in X and y
% Initialize some useful values
m = length(y); % number of training examples
% You need to return the following variables correctly
J = 0;
% ====================== YOUR CODE HERE ======================
% Instructions: Compute the cost of a particular choice of theta
% You should set J to the cost.
J = (1/(2*m))*sum(((X*theta) - y).^2)
% =========================================================================
end
where:
X is a 2 column matrix of m rows having all the elements of the first column set to the value 1:
X =
1.0000 6.1101
1.0000 5.5277
1.0000 8.5186
...... ......
...... ......
...... ......
y is a vector of m elements (as X):
y =
17.59200
9.13020
13.66200
........
........
........
Finnally theta is a 2 columns vector having 0 asvalues like this:
theta = zeros(2, 1); % initialize fitting parameters
theta
theta =
0
0
Ok, coming back to my working solution:
J = (1/(2*m))*sum(((X*theta) - y).^2)
specifically to this matrix multiplication (multiplication between the matrix X and the vector theta): I know that it is a valid matrix multiplication because the number of column of X (2 columns) is equal to the number of rows of theta (2 rows) so it is a perfectly valid matrix multiplication.
My doubt that is driving me crazy (probably it is a trivial doubt) is related to the previous course slide context:
As you can see in the second equation used to calculated the current h_theta(x) value it is using the transposed theta vector and not the theta vector as done in the code.
Why ?!?!
I suspect that it depends only on how was created the theta vector. It was build in this way:
theta = zeros(2, 1); % initialize fitting parameters
that is generating a 2 line 1 column vector instead of a classic one line 2 column vector. So maybe I have not to transpose it. But I am absolutely not sure about this assertion.
Is my intuition correct or what am I missing?
Your intuition is correct. Effectively it does not matter whether you perform the multiplication as theta.' * X or as X.' * theta, since this either generates a horizontal vector or a vertical vector of the hypothesis representing all observations, and what you're expected to do next is subtract the y vector from the hypothesis vector at each observation, and sum the results. So as long as y has the same orientation as your hypothesis and you subtract at each equivalent point, then the scalar end-result of the summation will be the same.
Often enough, you'll see the X.' * theta version preferred over theta.' * X purely for convenience, to avoid transposing over and over again just to be consistent with the mathematical notation. But this is fine, since the underlying math doesn't really change, only the order of equivalent operations.
I agree it's confusing though, both because it makes it harder to follow the formula when the code effectively looks like it's doing something else, and also since it messes with the usual convention that a vertical vector represents 'coordinates', and a horizontal vector represents observations. In such cases, especially in languages like matlab / octave where the orientation of a vector isn't explicitly defined in the variable's type, it is doubly important to document what you expect the inputs to represent, and preferably there should have been assert statements in the code confirming the input has been passed in the correct orientation. Clearly here they felt it wasn't necessary because this code is acting under controlled conditions in a predefined exercise environment anyway, but it would have been good practice to do so from a software engineering point of view.

Matlab Convolution regarding the conv() function and length()/size() function

I'm kind've new to Matlab and stack overflow to begin with, so if I do something wrong outside of the guidelines, please don't hesitate to point it out. Thanks!
I have been trying to do convolution between two functions and I have been having a hard time trying to get it to work.
t=0:.01:10;
h=exp(-t);
x=zeros(size(t)); % When I used length(t), I would get an error that says in conv(), A and B must be vectors.
x(1)=2;
x(4)=5;
y=conv(h,x);
figure; subplot(3,1,1);plot(t,x); % The discrete function would not show (at x=1 and x=4)
subplot(3,1,2);plot(t,h);
subplot(3,1,3);plot(t,y(1:length(t))); %Nothing is plotted here when ran
I commented my issues with the code. I don't understand the difference of length and size in this case and how it would make a difference.
For the second comment, x=1 should have an amplitude of 2. While x=4 should have an amplitude of 5. When plotted, it only shows nothing in the locations specified but looks jumbled up at x=0. I'm assuming that's the reason why the convoluted plot won't be displayed.
The original problem statement is given if it helps to understand what I was thinking throughout.
Consider an input signal x(t) that consists of two delta functions at t = 1 and t = 4 with amplitudes A1 = 5 and A2 = 2, respectively, to a linear system with impulse response h that is an exponential pulse (h(t) = e ^−t ). Plot x(t), h(t) and the output of the linear system y(t) for t in the range of 0 to 10 using increments of 0.01. Use the MATLAB built-in function conv.
The initial question regarding size vs length
length yields a scalar that is equal to the largest dimension of the input. In the case of your array, the size is 1 x N, so length yields N.
size(t)
% 1 1001
length(t)
% 1001
If you pass a scalar (N) to ones, zeros, or a similar function, it will create a square matrix that is N x N. This results in the error that you see when using conv since conv does not accept matrix inputs.
size(ones(length(t)))
% 1001 1001
When you pass a vector to ones or zeros, the output will be that size so since size returns a vector (as shown above), the output is the same size (and a vector) so conv does not have any issues
size(ones(size(t)))
% 1 1001
If you want a vector, you need to explicitly specify the number of rows and columns. Also, in my opinion, it's better to use numel to the number of elements in a vector as it's less ambiguous than length
z = zeros(1, numel(t));
The second question regarding the convolution output:
First of all, the impulses that you create are at the first and fourth index of x and not at the locations where t = 1 and t = 4. Since you create t using a spacing of 0.01, t(1) actually corresponds to t = 0 and t(4) corresponds to t = 0.03
You instead want to use the value of t to specify where to put your impulses
x(t == 1) = 2;
x(t == 4) = 5;
Note that due to floating point errors, you may not have exactly t == 1 and t == 4 so you can use a small epsilon instead
x(abs(t - 1) < eps) = 2;
x(abs(t - 4) < eps) = 5;
Once we make this change, we get the expected scaled and shifted versions of the input function.

Solving Least square using MATLAB

Assume we want to determine the coefficients of a polynomial equation that is approximating the tangent function between 0 to 1, as follow:
-A is m×n vandermonde matrix. The entries are populated using m value between 0 to 11(given as input).
-The corresponding vector b is calculated using tangent function.
-x is calculated by typing x= A\b in MATLAB.
Now, using MATLAB, the computed x are subsittued in Ax. The result is plotted and it is pretty close to tangent function. But if I use polyval function of n−1 degree (in MATLAB) to calculate b, the resulting plot is significantly different from the original b. I cannot understand the reason for such a significant difference between the results of these two methods.
Here is the code:
clear all;
format long;
m = 60;
n = 11;
t = linspace(0,1,m);
A= fliplr(vander(t));
A=A(:,1:n);
b=tan(t');
x= A\b;
y=polyval(x, t);
plot(t,y,'r')
y2= A*x
hold on;
plot(t,y2,'g.');
hold on;
plot(t,tan(t),'--b');
Any insight would be appreciated. Thank you.
After A= fliplr(vander(t)) the A matrix is equal to
1 t(1) t(1)^2 ...
1 t(2) t(2)^2 ...
...
1 t(m) t(m)^2 ...
It is not correct because polyval accepts the coefficients in descending powers. You don't need to flip the columns of A:
A= vander(t);
A= A(:,end-n+1:end);

Convolution of multiple 1D signals in a 2D matrix with multiple 1D kernels in a 2D matrix

I have a randomly defined H matrix of size 600 x 10. Each element in this matrix H can be represented as H(k,t). I obtained a speech spectrogram S which is 600 x 597. I obtained it using Mel features, so it should be 40 x 611 but then I used a frame stacking concept in which I stacked 15 frames together. Therefore it gave me (40x15) x (611-15+1) which is 600 x 597.
Now I want to obtain an output matrix Y which is given by the equation based on convolution Y(k,t) = ∑ H(k,τ)S(k,t-τ). The sum goes from τ=0 to τ=Lh-1. Lh in this case would be 597.
I don't know how to obtain Y. Also, my doubt is the indexing into both H and S when computing the convolution. Specifically, for Y(1,1), we have:
Y(1,1) = H(1,0)S(1,1) + H(1,1)S(1,0) + H(1,2)S(1,-1) + H(1,3)S(1,-2) + ...
Now, there is no such thing as negative indices in MATLAB - for example, S(1,-1) S(1,-2) and so on. So, what type of convolution should I use to obtain Y? I tried using conv2 or fftfilt but I think that will not give me Y because Y must also be the size of S.
That's very easy. That's a convolution on a 2D signal only being applied to 1 dimension. If we assume that the variable k is used to access the rows and t is used to access the columns, you can consider each row of H and S as separate signals where each row of S is a 1D signal and each row of H is a convolution kernel.
There are two ways you can approach this problem.
Time domain
If you want to stick with time domain, the easiest thing would be to loop over each row of the output, find the convolution of each pair of rows of S and H and store the output in the corresponding output row. From what I can tell, there is no utility that can convolve in one dimension only given an N-D signal.... unless you go into frequency domain stuff, but let's leave that for later.
Something like:
Y = zeros(size(S));
for idx = 1 : size(Y,1)
Y(idx,:) = conv(S(idx,:), H(idx,:), 'same');
end
For each row of the output, we perform a row-wise convolution with a row of S and a row of H. I use the 'same' flag because the output should be the same size as a row of S... which is the bigger row.
Frequency domain
You can also perform the same computation in frequency domain. If you know anything about the properties of convolution and the Fourier Transform, you know that convolution in time domain is multiplication in the frequency domain. You take the Fourier Transform of both signals, multiply them element-wise, then take the Inverse Fourier Transform back.
However, you need to keep the following intricacies in mind:
Performing a full convolution means that the final length of the output signal is length(A)+length(B)-1, assuming A and B are 1D signals. Therefore, you need to make sure that both A and B are zero-padded so that they both match the same size. The reason why you make sure that the signals are the same size is to allow for the multiplication operation to work.
Once you multiply the signals in the frequency domain then take the inverse, you will see that each row of Y is the full length of the convolution. To ensure that you get an output that is the same size as the input, you need to trim off some points at the beginning and at the end. Specifically, since each kernel / column length of H is 10, you would have to remove the first 5 and last 5 points of each signal in the output to match what you get in the for loop code.
Usually after the inverse Fourier Transform, there are some residual complex coefficients due to the nature of the FFT algorithm. It's good practice to use real to remove the complex valued parts of the results.
Putting all of this theory together, this is what the code would look like:
%// Define zero-padded H and S matrices
%// Rows are the same, but columns must be padded to match point #1
H2 = zeros(size(H,1), size(H,2)+size(S,2)-1);
S2 = zeros(size(S,1), size(H,2)+size(S,2)-1);
%// Place H and S at the beginning and leave the rest of the columns zero
H2(:,1:size(H,2)) = H;
S2(:,1:size(S,2)) = S;
%// Perform Fourier Transform on each row separately of padded matrices
Hfft = fft(H2, [], 2);
Sfft = fft(S2, [], 2);
%// Perform convolution
Yfft = Hfft .* Sfft;
%// Take inverse Fourier Transform and convert to real
Y2 = real(ifft(Yfft, [], 2));
%// Trim off unnecessary values
Y2 = Y2(:,size(H,2)/2 + 1 : end - size(H,2)/2 + 1);
Y2 should be the convolved result and should match Y in the previous for loop code.
Comparison between them both
If you actually want to compare them, we can. What we'll need to do first is define H and S. To reconstruct what I did, I generated random values with a known seed:
rng(123);
H = rand(600,10);
S = rand(600,597);
Once we run the above code for both the time domain version and frequency domain version, let's see how they match up in the command prompt. Let's show the first 5 rows and 5 columns:
>> format long g;
>> Y(1:5,1:5)
ans =
1.63740867892464 1.94924208172753 2.38365646354643 2.05455605619097 2.21772526557861
2.04478411247085 2.15915645246324 2.13672842742653 2.07661341840867 2.61567534623066
0.987777477630861 1.3969752201781 2.46239452105228 3.07699790208937 3.04588738611503
1.36555260994797 1.48506871890027 1.69896157726456 1.82433906982894 1.62526864072424
1.52085236885395 2.53506897420001 2.36780282057747 2.22335617436888 3.04025523335182
>> Y2(1:5,1:5)
ans =
1.63740867892464 1.94924208172753 2.38365646354643 2.05455605619097 2.21772526557861
2.04478411247085 2.15915645246324 2.13672842742653 2.07661341840867 2.61567534623066
0.987777477630861 1.3969752201781 2.46239452105228 3.07699790208937 3.04588738611503
1.36555260994797 1.48506871890027 1.69896157726456 1.82433906982894 1.62526864072424
1.52085236885395 2.53506897420001 2.36780282057747 2.22335617436888 3.04025523335182
Looks good to me! As another measure, let's figure out what the largest difference is between one value in Y and a corresponding value in Y2:
>> max(abs(Y(:) - Y2(:)))
ans =
5.32907051820075e-15
That's saying that the max error seen between both outputs is in the order of 10-15. I'd say that's pretty good.

Numerical derivative of a vector

I have a problem with numerical derivative of a vector that is x: Nx1 with respect to another vector t (time) that is the same size of x.
I do the following (x is chosen to be sine function as an example):
t=t0:ts:tf;
x=sin(t);
xd=diff(x)/ts;
but the answer xd is (N-1)x1 and I figured out that it does not compute derivative corresponding to the first element of x.
is there any other way to compute this derivative?
You are looking for the numerical gradient I assume.
t0 = 0;
ts = pi/10;
tf = 2*pi;
t = t0:ts:tf;
x = sin(t);
dx = gradient(x)/ts
The purpose of this function is a different one (vector fields), but it offers what diff doesn't: input and output vector of equal length.
gradient calculates the central difference between data points. For an
array, matrix, or vector with N values in each row, the ith value is
defined by
The gradient at the end points, where i=1 and i=N, is calculated with
a single-sided difference between the endpoint value and the next
adjacent value within the row. If two or more outputs are specified,
gradient also calculates central differences along other dimensions.
Unlike the diff function, gradient returns an array with the same
number of elements as the input.
I know I'm a little late to the game here, but you can also get an approximation of the numerical derivative by taking the derivatives of the polynomial (cubic) splines that runs through your data:
function dy = splineDerivative(x,y)
% the spline has continuous first and second derivatives
pp = spline(x,y); % could also use pp = pchip(x,y);
[breaks,coefs,K,r,d] = unmkpp(pp);
% pre-allocate the coefficient vector
dCoeff = zeroes(K,r-1);
% Columns are ordered from highest to lowest power. Both spline and pchip
% return 4xn matrices, ordered from 3rd to zeroth power. (Thanks to the
% anonymous person who suggested this edit).
dCoeff(:, 1) = 3 * coefs(:, 1); % d(ax^3)/dx = 3ax^2;
dCoeff(:, 2) = 2 * coefs(:, 2); % d(ax^2)/dx = 2ax;
dCoeff(:, 3) = 1 * coefs(:, 3); % d(ax^1)/dx = a;
dpp = mkpp(breaks,dCoeff,d);
dy = ppval(dpp,x);
The spline polynomial is always guaranteed to have continuous first and second derivatives at each point. I haven not tested and compared this against using pchip instead of spline, but that might be another option as it too has continuous first derivatives (but not second derivatives) at every point.
The advantage of this is that there is no requirement that the step size be even.
There are some options to work-around your issue.
First: you can make your domain larger. Instead of N, use N+1 gridpoints.
Second: depending on the end-point of interest, you can use
Forward difference: F(x + dx) - F(x)
Backward difference: F(x) - F(x - dx)