I have a matrix of 50-by-1 that is demodulated data. As this matrix has only one element in each row, I want to repeat this only element 16 times in each row so the matrix become 50 by 16. I did it using the repmat(A,16) command in Matlab. Now at receiving end noise is also added in matrix of 50 by 16. I want to get it back of 50 by 1 matrix. How can I do this?
I tried averaging of all rows but it is not a valid method. How can I know when an error is occurring in this process?
You are describing a problem of the form y = A * x + n, where y is the observed data, A is a known linear transform, and n is noise. The least squares estimate is the simplest estimate of the unknown vector x. The keys here are to express the repmat() function as a matrix and the observed data as a vector (i.e., a 50*16x1 vector rather than a 50x16 matrix).
x = 10 * rand(50,1); % 50x1 data vector;
A = repmat(eye(length(x)),[16,1]); % This stacks 16 replicas of x.
n = rand(50*16,1); % Noise
y = A * x + n; % Observed data
xhat = A \ y; % Least squares estimate of x.
As for what the inverse (what I assume you mean by 'reverse') of A is, it doesn't have one. If you look at its rank, you'll see it is only 50. The best you can do is to use its pseudoinverse, which is what the \ operator does.
I hope the above helps.
Related
I know that ifft sums multiple sine waves up from data obtain from doing an fft on a signal. is there a way to do a ifft using square waves instead of sine waves?
I'm not trying to get the original signal back but trying to rebuild it using square waves from the data taken from the fft instead of the normal sine wave summation process.
See simple example below: the signals I will be using are human audio signals about 60 seconds long so I'm trying to see if I can use / alter the ifft command in some way.
PS: I'm using Octave 4.0 which is similar to Matlab
clear all,clf reset, clc,tic
Fs = 200; % Sampling frequency
t=linspace(0,1,Fs);
freq=2;
%1 create signal
ya = .5*sin(freq*pi*2*t+pi);
%2 create frequency domain
ya_fft = fft(ya);
%3 rebuild signal
mag = abs(ya_fft);
phase = unwrap(angle(ya_fft));
ya_newifft=ifft(mag.*exp(i*phase));
ifft_sig_combined_L1=ifft(mag.*exp(i*phase),Fs); %use Fs to get correct file length
% square wave
vertoffset=0.5;
A=1
T = 1/freq; % period of the signal
square = mod(t * A / T, A) > A / 2;
square = square - vertoffset;
subplot(3,1,1);
plot(t,ya,'r')
title('orignal signal')
subplot(3,1,2);
plot(t,ifft_sig_combined_L1)
title('rebuilt signal')
subplot(3,1,3);
plot(t,square)
title('rebuilt signal with square wave')
Define the basis vectors you want to use and let them be the columns of a matrix, A. If b is your signal, then just get the least squares solution to Ax = b. If A is full rank, then you will be able to represent b exactly.
Edit:
Think about what a matrix-vector product does: Each column of the matrix is multiplied by the corresponding element of the vector (i.e., the n^th column of the matrix is multiplied by the n^th element of the vector) and the resulting products are summed together. (This would be a lot easier to illustrate if this site supported latex.) In Matlab, a horrible but hopefully illustrative way to do this is
A = some_NxN_matrix;
x = some_Nx1_vector;
b = zeros( size(A,1), 1 );
for n = 1 : length(x)
b = b + A(:,n) * x(n);
end
(Of course, you would never actually do the above but rather b = A*x;.)
Now define whatever square waves you want to use and assign each to its own Nx1 vector. Call these vectors s_1, s_2, ..., s_M, where M is the number of square waves you are using. Now let
A = [s1, s2, ..., s_M];
According to your question, you want to represent your signal as a weighted sum of these square waves. (Note that this is exactly what a DFT does it just uses orthogonal sinusoids rather than square waves.) To weight and sum these square waves, all you have to do is find the matrix-vector product A*x, where x is the vector of coefficients that weight each column (see the above paragraph). Now, if your signal is b and you want to the find the x that will best sum the square waves in order to approximate b, then all you have to do is solve A*x=b. In Matlab, this is given by
x = A \ b;
The rest is just linear algebra. If a left-inverse of A exists (i.e., if A has dimensions M x N and rank N, with M > N), then (A^-1) * A is an identity matrix and
(A^-1) * A * x = (A^-1) * b,
which implies that x = (A^-1) * b, which is what x = A \ b; will return in Matlab. If A has dimensions M x N and rank M, with N > M, then the system is underdetermined and a left-inverse does not exist. In this case you have to use the psuedo-inverse to solve the system. Now suppose that A is NxN with rank N, so that both the left- and right-inverse exist. In this case, x will give an exact representation of b:
x = (A^-1) * b
A * x = A * (A^-1) * b = b
If you want an example of A that uses square waves to get an exact representation of the input signal, check out the Haar transform. There is a function available here.
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.
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.
I'm trying to reconstruct a 3d image from two calibrated cameras. One of the steps involved is to calculate the 3x3 essential matrix E, from two sets of corresponding (homogeneous) points (more than the 8 required) P_a_orig and P_b_orig and the two camera's 3x3 internal calibration matrices K_a and K_b.
We start off by normalizing our points with
P_a = inv(K_a) * p_a_orig
and
P_b = inv(K_b) * p_b_orig
We also know the constraint
P_b' * E * P_a = 0
I'm following it this far, but how do you actually solve that last problem, e.g. finding the nine values of the E matrix? I've read several different lecture notes on this subject, but they all leave out that crucial last step. Likely because it is supposedly trivial math, but I can't remember when I last did this and I haven't been able to find a solution yet.
This equation is actually pretty common in geometry algorithms, essentially, you are trying to calculate the matrix X from the equation AXB=0. To solve this, you vectorise the equation, which means,
vec() means vectorised form of a matrix, i.e., simply stack the coloumns of the matrix one over the another to produce a single coloumn vector. If you don't know the meaning of the scary looking symbol, its called Kronecker product and you can read it from here, its easy, trust me :-)
Now, say I call the matrix obtained by Kronecker product of B^T and A as C.
Then, vec(X) is the null vector of the matrix C and the way to obtain that is by doing the SVD decomposition of C^TC (C transpose multiplied by C) and take the the last coloumn of the matrix V. This last coloumn is nothing but your vec(X). Reshape X to 3 by 3 matrix. This is you Essential matrix.
In case you find this maths too daunting to code, simply use the following code by Y.Ma et.al:
% p are homogenius coordinates of the first image of size 3 by n
% q are homogenius coordinates of the second image of size 3 by n
function [E] = essentialDiscrete(p,q)
n = size(p);
NPOINTS = n(2);
% set up matrix A such that A*[v1,v2,v3,s1,s2,s3,s4,s5,s6]' = 0
A = zeros(NPOINTS, 9);
if NPOINTS < 9
error('Too few mesurements')
return;
end
for i = 1:NPOINTS
A(i,:) = kron(p(:,i),q(:,i))';
end
r = rank(A);
if r < 8
warning('Measurement matrix rank defficient')
T0 = 0; R = [];
end;
[U,S,V] = svd(A);
% pick the eigenvector corresponding to the smallest eigenvalue
e = V(:,9);
e = (round(1.0e+10*e))*(1.0e-10);
% essential matrix
E = reshape(e, 3, 3);
You can do several things:
The Essential matrix can be estimated using the 8-point algorithm, which you can implement yourself.
You can use the estimateFundamentalMatrix function from the Computer Vision System Toolbox, and then get the Essential matrix from the Fundamental matrix.
Alternatively, you can calibrate your stereo camera system using the estimateCameraParameters function in the Computer Vision System Toolbox, which will compute the Essential matrix for you.
I try to write an algorithm which determine $\mu$, $\sigma$,$\pi$ for each class from a mixture multivariate normal distribution.
I finish with the algorithm partially, it works when I set the random guess values($\mu$, $\sigma$,$\pi$) near from the real value. But when I set the values far from the real one, the algorithm does not converge. The sigma goes to 0 $(2.30760684053766e-24 2.30760684053766e-24)$.
I think the problem is my covarience calculation, I am not sure that this is the right way. I found this on wikipedia.
I would be grateful if you could check my algorithm. Especially the covariance part.
Have a nice day,
Thanks,
2 mixture gauss
size x = [400, 2] (400 point 2 dimension gauss)
mu = 2 , 2 (1 row = first gauss mu, 2 row = second gauss mu)
for i = 1 : k
gaussEvaluation(i,:) = pInit(i) * mvnpdf(x,muInit(i,:), sigmaInit(i, :) * eye(d));
gaussEvaluationSum = sum(gaussEvaluation(i, :));
%mu calculation
for j = 1 : d
mu(i, j) = sum(gaussEvaluation(i, :) * x(:, j)) / gaussEvaluationSum;
end
%sigma calculation methode 1
%for j = 1 : n
% v = (x(j, :) - muNew(i, :));
% sigmaNew(i) = sigmaNew(i) + gaussEvaluation(i,j) * (v * v');
%end
%sigmaNew(i) = sigmaNew(i) / gaussEvaluationSum;
%sigma calculation methode 2
sub = bsxfun(#minus, x, mu(i,:));
sigma(i,:) = sum(gaussEvaluation(i,:) * (sub .* sub)) / gaussEvaluationSum;
%p calculation
p(i) = gaussEvaluationSum / n;
Two points: you can observe this even when you implement gaussian mixture EM correctly, but in your case, the code does seem to be incorrect.
First, this is just a problem that you have to deal with when fitting mixtures of gaussians. Sometimes one component of the mixture can collapse on to a single point, resulting in the mean of the component becoming that point and the variance becoming 0; this is known as a 'singularity'. Hence, the likelihood also goes to infinity.
Check out slide 42 of this deck: http://www.cs.ubbcluj.ro/~csatol/gep_tan/Bishop-CUED-2006.pdf
The likelihood function that you are evaluating is not log-concave, so the EM algorithm will not converge to the same parameters with different initial values. The link I gave above also gives some solutions to avoid this over-fitting problem, such as putting a prior or regularization term on your parameters. You can also consider running multiple times with different starting parameters and discarding any results with variance 0 components as having over-fitted, or just reduce the number of components you are using.
In your case, your equation is right; the covariance update calculation on Wikipedia is the same as the one on slide 45 of the above link. However, if you are in a 2d space, for each component the mean should be a length 2 vector and the covariance should be a 2x2 matrix. Hence your code (for two components) is wrong because you have a 2x2 matrix to store the means and a 2x2 matrix to store the covariances; it should be a 2x2x2 matrix.