I've read some explanations of how autocorrelation can be more efficiently calculated using the fft of a signal, multiplying the real part by the complex conjugate (Fourier domain), then using the inverse fft, but I'm having trouble realizing this in Matlab because at a detailed level.
Just like you stated, take the fft and multiply pointwise by its complex conjugate, then use the inverse fft (or in the case of cross-correlation of two signals: Corr(x,y) <=> FFT(x)FFT(y)*)
x = rand(100,1);
len = length(x);
%# autocorrelation
nfft = 2^nextpow2(2*len-1);
r = ifft( fft(x,nfft) .* conj(fft(x,nfft)) );
%# rearrange and keep values corresponding to lags: -(len-1):+(len-1)
r = [r(end-len+2:end) ; r(1:len)];
%# compare with MATLAB's XCORR output
all( (xcorr(x)-r) < 1e-10 )
In fact, if you look at the code of xcorr.m, that's exactly what it's doing (only it has to deal with all the cases of padding, normalizing, vector/matrix input, etc...)
By the Wiener–Khinchin theorem, the power-spectral density (PSD) of a function is the Fourier transform of the autocorrelation. For deterministic signals, the PSD is simply the magnitude-squared of the Fourier transform. See also the convolution theorem.
When it comes to discrete Fourier transforms (i.e. using FFTs), you actually get the cyclic autocorrelation. In order to get proper (linear) autocorrelation, you must zero-pad the original data to twice its original length before taking the Fourier transform. So something like:
x = [ ... ];
x_pad = [x zeros(size(x))];
X = fft(x_pad);
X_psd = abs(X).^2;
r_xx = ifft(X_psd);
Related
I have a data which contains 16 elements:
x=[8.57837e-08, 2.07482e-06, 4.43796e-06, 7.66462e-06, 1.10232e-05, 1.35811e-05, 1.27958e-05, 5.94217e-06, 2.49168e-08, -6.58389e-06, -1.30551e-05, -1.345e-05, -1.07471e-05, -7.38637e-06, -4.42876e-06, -1.88811e-06 ];
A = length(x)
I do DTFT-DFT like dirac signals:
n=0:A;
syms w
X_w=0;
for i=1:length(x)
X_w=X_w+x(i)*exp(-j*w*n(i));
end
figure;fplot(angle(X_w),[0 2*pi]),title('DTFT phase graph')
figure;fplot(abs(X_w),[0 2*pi]),title('DTFT amplitude graph')
hold on
%DFT
N=50;
k=0:N-1;
DFT_X=[];
for k=0:N-1
Xk=0;
for i=1:length(x)
Xk=Xk+x(i).*exp(-j*(2*pi/N).*k.*n(i));
end
DFT_X=[DFT_X Xk];
end
w=2*pi/N*(0:N-1);
stem(w,abs(DFT_X))`
The problem is I want to write this signal with cosinus and sinus components. But I don't really know how can I do.
Thank you all,
Emre.
Direct computation of the Fourier coefficients might be a better option than trying to relate the DFT to the DFS. Looking at the continuous time formulas:
all you'd need to do is sum the input signal x multiplied (element wise) by a cosine over it's domain for the real coefficient and sum the input signal x multiplied (element wise) by a sine over its domain for the imaginary coefficients.
Secondly, you could potentially use conjugate symmetry and these formulas to calculate the relationship between your Xk and the desired An and Bn coefficients.
Xk = (An-j*Bn)/2
and
Xk* = (An+j*Bn)/2)
I have a question about the sgolay function in Matlab R2013a. My database has 165 spectra with 2884 variables and I would like to take the first and second derivatives of them. How might I define the inputs K and F to sgolay?
Below is an example:
sgolay is used to smooth a noisy sinusoid and compare the resulting first and second derivatives to the first and second derivatives computed using diff. Notice how using diff amplifies the noise and generates useless results.
K = 4; % Order of polynomial fit
F = 21; % Window length
[b,g] = sgolay(K,F); % Calculate S-G coefficients
dx = .2;
xLim = 200;
x = 0:dx:xLim-1;
y = 5*sin(0.4*pi*x)+randn(size(x)); % Sinusoid with noise
HalfWin = ((F+1)/2) -1;
for n = (F+1)/2:996-(F+1)/2,
% Zero-th derivative (smoothing only)
SG0(n) = dot(g(:,1), y(n - HalfWin: n + HalfWin));
% 1st differential
SG1(n) = dot(g(:,2), y(n - HalfWin: n + HalfWin));
% 2nd differential
SG2(n) = 2*dot(g(:,3)', y(n - HalfWin: n + HalfWin))';
end
SG1 = SG1/dx; % Turn differential into derivative
SG2 = SG2/(dx*dx); % and into 2nd derivative
% Scale the "diff" results
DiffD1 = (diff(y(1:length(SG0)+1)))/ dx;
DiffD2 = (diff(diff(y(1:length(SG0)+2)))) / (dx*dx);
subplot(3,1,1);
plot([y(1:length(SG0))', SG0'])
legend('Noisy Sinusoid','S-G Smoothed sinusoid')
subplot(3, 1, 2);
plot([DiffD1',SG1'])
legend('Diff-generated 1st-derivative', 'S-G Smoothed 1st-derivative')
subplot(3, 1, 3);
plot([DiffD2',SG2'])
legend('Diff-generated 2nd-derivative', 'S-G Smoothed 2nd-derivative')
Taking derivatives in an inherently noisy process. Thus, if you already have some noise in your data, indeed, it will be magnified as you take higher order derivatives. Savitzky-Golay is a very useful way of combining smoothing and differentiation into one operation. It's a general method and it computes derivatives to an arbitrary order. There are trade-offs, though. Other special methods exist for data with a certain structure.
In terms of your application, I don't have any concrete answers. Much depends on the nature of the data (sampling rate, noise ratio, etc.). If you use too much smoothing, you'll smear your data or produce aliasing. Same thing if you over-fit the data by using high order polynomial coefficients, K. In your demo code you should also plot the analytical derivatives of the sin function. Then play with different amounts of input noise and smoothing filters. Such a tool with known exact answers may be helpful if you can approximate aspects of your real data. In practice, I try to use as little smoothing as possible in order to produce derivatives that aren't too noisy. Often this means a third-order polynomial (K = 3) and a window size, F, as small as possible.
So yes, many suggest that you use your eyes to tune these parameters. However, there has also been some very recent research on choosing the coefficients automatically: On the Selection of Optimum Savitzky-Golay Filters (2013). There are also alternatives to Savitzky-Golay, e.g., this paper based on regularization, but you may need to implement them yourself in Matlab.
By the way, a while back I wrote a little replacement for sgolay. Like you, I only needed the second output, the differentiation filters, G, so that's all it calculates. This function is also faster (by about 2–4 times):
function G=sgolayfilt(k,f)
%SGOLAYFILT Savitzky-Golay differentiation filters
s = vander(0.5*(1-f):0.5*(f-1));
S = s(:,f:-1:f-k);
[~,R] = qr(S,0);
G = S/R/R';
A full version of this function with input validation is available on my GitHub.
I'm trying to get wavelet decomposition of arcsin(x) using, say, Haar wavelets
When using both Matlab's dwt or wavedec functions, I get strange values for approximating coefficients. Since applying low-pass Haar wavelets's filter equals to performing half-sum and the maximum of arcsin is pi/2, I assume that approximating coefficients can't surpass pi/2, yet this code:
x = linspace(0,1,128);
y = asin(x);
[cA, cD] = dwt(y, 'haar'); %//cA for approximating coefficients
returns values more than pi/2 in cA. Why is that?
I believe what makes you confused here is thinking that Haar's filter just averages two adjacent numbers when computing 1-level approximation coefficients. Due to the energy preservation feature of the scaling function, each pair of numbers gets divided by sqrt(2) instead of 2. In fact, you could see what a particular wavelet filter does by typing in the following command (for the Haar filter in this case):
[F1,F2] = wfilters('haar','d')
F1 =
0.7071 0.7071
F2 =
-0.7071 0.7071
You can then check the validity of what you have gotten above by constructing a simple loop:
CA_compare = zeros(1,64);
for k = 1 : 64
CA_compare(k) = dot( y(2*k-1 : 2*k), F1 );
end
You will then see that "CA_compare" contains exactly the same values as your "cA" does.
Hope this helps.
I was aquainted in using the fft in matlab with the code
fft(signal,[],n)
where n tells the dimension on which to apply the fft as from Matlab documentation:
http://www.mathworks.it/it/help/matlab/ref/fft.html
I would like to do the same with dct.
Is this possible? I could not find any useful information around.
Thanks for the help.
Luigi
dct does not have the option to pick a dimension like fft. You will have to either transpose your input to operate on rows or pick one vector from your signal and operate on that.
yep!
try it:
matrix = dctmtx(n);
signal_dct = matrix * signal;
Edit
Discrete cosine transform.
Y = dct(X) returns the discrete cosine transform of X.
The vector Y is the same size as X and contains the discrete cosine transform coefficients.
Y = dct(X,N) pads or truncates the vector X to length N before transforming.
If X is a matrix, the dct operation is applied to each column. This transform can be inverted using IDCT.
% Example:
% Find how many dct coefficients represent 99% of the energy
% in a sequence.
x = (1:100) + 50*cos((1:100)*2*pi/40); % Input Signal
X = dct(x); % Discrete cosine transform
[XX,ind] = sort(abs(X)); ind = fliplr(ind);
num_coeff = 1;
while (norm([X(ind(1:num_coeff)) zeros(1,100-num_coeff)])/norm(X)<.99)
num_coeff = num_coeff + 1;
end;
num_coeff
I have an assignment to implement a Ram-Lak filter, but nearly no information given on it (except look at fft, ifft, fftshift, ifftshift).
I have a sinogram that I have to filter via Ram-Lak. Also the number of projections is given.
I try to use the filter
1/4 if I == 0
(b^2)/(2*pi^2) * 0 if I even
-1/(pi^2 * I^2) if I odd
b seems to be the cut-off frequency, I has something to do with the sampling rate?
Also it is said that the convolution of two functions is a simple multiplication in Fourier space.
I do not understand how to implement the filter at all, especially with no b given, not told what I is and no idea how to apply this to the sinogram, I hope someone can help me here. I spent 2hrs googling and trying to understand what is needed to do here, but I could not understand how to implement it.
The formula you listed is an intermediate result if you wanted to do an inverse Radon transform without filtering in the Fourier domain. An alternative is to do the entire filtered back projection algorithm using convolution in the spatial domain, which might have been faster 40 years ago; you would eventually rederive the formula you posted. However, I wouldn't recommended it now, especially not for your first reconstruction; you should really understand the Hilbert transform first.
Anyway, here's some Matlab code which does the obligatory Shepp-Logan phantom filtered back projection reconstruction. I show how you can do your own filtering with the Ram-Lak filter. If I was really motivated, I would replace radon/iradon with some interp2 commands and summations.
phantomData=phantom();
N=size(phantomData,1);
theta = 0:179;
N_theta = length(theta);
[R,xp] = radon(phantomData,theta);
% make a Ram-Lak filter. it's just abs(f).
N1 = length(xp);
freqs=linspace(-1, 1, N1).';
myFilter = abs( freqs );
myFilter = repmat(myFilter, [1 N_theta]);
% do my own FT domain filtering
ft_R = fftshift(fft(R,[],1),1);
filteredProj = ft_R .* myFilter;
filteredProj = ifftshift(filteredProj,1);
ift_R = real(ifft(filteredProj,[],1));
% tell matlab to do inverse FBP without a filter
I1 = iradon(ift_R, theta, 'linear', 'none', 1.0, N);
subplot(1,3,1);imagesc( real(I1) ); title('Manual filtering')
colormap(gray(256)); axis image; axis off
% for comparison, ask matlab to use their Ram-Lak filter implementation
I2 = iradon(R, theta, 'linear', 'Ram-Lak', 1.0, N);
subplot(1,3,2);imagesc( real(I2) ); title('Matlab filtering')
colormap(gray(256)); axis image; axis off
% for fun, redo the filtering wrong on purpose
% exclude high frequencies to create a low-resolution reconstruction
myFilter( myFilter > 0.1 ) = 0;
ift_R = real(ifft(ifftshift(ft_R .* myFilter,1),[],1));
I3 = iradon(ift_R, theta, 'linear', 'none', 1.0, N);
subplot(1,3,3);imagesc( real(I3) ); title('Low resolution filtering')
colormap(gray(256)); axis image; axis off