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
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 have a problem when calculate discrete Fourier transform in MATLAB, apparently get the right result but when plot the amplitude of the frequencies obtained you can see values very close to zero which should be exactly zero. I use my own implementation:
function [y] = Discrete_Fourier_Transform(x)
N=length(x);
y=zeros(1,N);
for k = 1:N
for n = 1:N
y(k) = y(k) + x(n)*exp( -1j*2*pi*(n-1)*(k-1)/N );
end;
end;
end
I know it's better to use fft of MATLAB, but I need to use my own implementation as it is for college.
The code I used to generate the square wave:
x = [ones(1,8), -ones(1,8)];
for i=1:63
x = [x, ones(1,8), -ones(1,8)];
end
MATLAB version: R2013a(8.1.0.604) 64 bits
I have tried everything that has happened to me but I do not have much experience using MATLAB and I have not found information relevant to this issue in forums. I hope someone can help me.
Thanks in advance.
This will be a numerical problem. The values are in the range of 1e-15, while the DFT of your signal has values in the range of 1e+02. Most likely this won't lead to any errors when doing further processing. You can calculate the total squared error between your DFT and the MATLAB fft function by
y = fft(x);
yh = Discrete_Fourier_Transform(x);
sum(abs(yh - y).^2)
ans =
3.1327e-20
which is basically zero. I would therefore conclude: your DFT function works just fine.
Just one small remark: You can easily vectorize the DFT.
n = 0:1:N-1;
k = 0:1:N-1;
y = exp(-1j*2*pi/N * n'*k) * x(:);
With n'*k you create a matrix with all combinations of n and k. You then take the exp(...) of each of those matrix elements. With x(:) you make sure x is a column vector, so you can do the matrix multiplication (...)*x which automatically sums over all k's. Actually, I just notice, this is exactly the well-known matrix form of the DFT.
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)
I have the following code for calculating the result of a linear combination of Gaussian functions. What I'd really like to do is to vectorize this somehow so that it's far more performant in Matlab.
Note that y is a column vector (output), x is a matrix where each column corresponds to a data point and each row corresponds to a dimension (i.e. 2 rows = 2D), variance is a double, gaussians is a matrix where each column is a vector corresponding to the mean point of the gaussian and weights is a row vector of the weights in front of each gaussian. Note that the length of weights is 1 bigger than gaussians as weights(1) is the 0th order weight.
function [ y ] = CalcPrediction( gaussians, variance, weights, x )
basisFunctions = size(gaussians, 2);
xvalues = size(x, 2);
if length(weights) ~= basisFunctions + 1
ME = MException('TRAIN:CALC', 'The number of weights should be equal to the number of basis functions plus one');
throw(ME);
end
y = weights(1) * ones(xvalues, 1);
for xIdx = 1:xvalues
for i = 1:basisFunctions
diff = x(:, xIdx) - gaussians(:, i);
y(xIdx) = y(xIdx) + weights(i+1) * exp(-(diff')*diff/(2*variance));
end
end
end
You can see that at the moment I simply iterate over the x vectors and then the gaussians inside 2 for loops. I'm hoping that this can be improved - I've looked at meshgrid but that seems to only apply to vectors (and I have matrices)
Thanks.
Try this
diffx = bsxfun(#minus,x,permute(gaussians,[1,3,2])); % binary operation with singleton expansion
diffx2 = squeeze(sum(diffx.^2,1)); % dot product, shape is now [XVALUES,BASISFUNCTIONS]
weight_col = weights(:); % make sure weights is a column vector
y = exp(-diffx2/2/variance)*weight_col(2:end); % a column vector of length XVALUES
Note, I changed diff to diffx since diff is a builtin. I'm not sure this will improve performance as allocating arrays will offset increase by vectorization.
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);