Here's my attempt in implementing this lovely formula.
http://dl.dropbox.com/u/7348856/Picture1.png
%WIGNER Computes Wigner-Distribution on an image (difference of two images).
function[wd] = wigner(difference)
%Image size
[M, N, ~] = size(difference);
%Window size (5 x 5)
Md = 5;
Nd = 5;
%Fourier Transform
F = fft2(difference);
%Initializing the wigner picture
wd = zeros(M, N, 'uint8');
lambda =0.02;
value = (4/(Md*Nd));
for x = 1+floor(Md/2):M - floor(Md/2)
for y = 1+floor(Nd/2):N - floor(Nd/2)
for l = -floor(Nd/2) : floor(Nd/2)
for k = -floor(Md/2) : floor(Md/2)
kernel = exp(-lambda * norm(k,l));
kernel = kernel * value;
theta = 4 * pi * ((real(F(x, y)) * (k/M) )+ (imag(F(x, y)) * (l/N)));
wd(x, y) = (wd(x, y)) + (cos(theta) * difference(x + k, y + l) * difference(x - k, y - l) * (kernel));
end
end
end
end
end
As you can see, the outer two loops are for the sliding window, while the remaining inner ones are for the variables of the summation.
Now, my request for you my beloved stackoverflow users is: Can you help me improve these very nasty for loops that take more than its share of time, and turn it into vectorized loops?
And will that improvement be of a significant change?
Thank you.
this might not be what you are asking, but it seems (at first glance) that the order of the summations are independent and that instead of {x,y,l,k} you could go {l,k,x,y}. doing this will allow you to evaluate kernel fewer times by keeping it in the outer most loop.
Those four nested loops are basically processing each pixel in the image in a sliding-neighborhood style. I immediately thought of NLFILTER and IM2COL functions.
Here is my attempt at vectorizing the code. Note that I haven't thoroughly tested it, or compared performance against loop-based solution:
function WD = wigner(D, Md, Nd, lambda)
%# window size and lambda
if nargin<2, Md = 5; end
if nargin<3, Nd = 5; end
if nargin<4, lambda = 5; end
%# image size
[M,N,~] = size(D);
%# kernel = exp(-lambda*norm([k,l])
[K,L] = meshgrid(-floor(Md/2):floor(Md/2), -floor(Nd/2):floor(Nd/2));
K = K(:); L = L(:);
kernel = exp(-lambda .* sqrt(K.^2+L.^2));
%# frequency-domain part
F = fft2(D);
%# f(x+k,y+l) * f(x-k,y-l) * kernel
C = im2col(D, [Md Nd], 'sliding');
X1 = bsxfun(#times, C .* flipud(C), kernel);
%# cos(theta)
C = im2col(F, [Md Nd], 'sliding');
C = C(round(Md*Nd/2),:); %# take center pixels
theta = bsxfun(#times, real(C), K/M) + bsxfun(#times, imag(C), L/N);
X2 = cos(4*pi*theta);
%# combine both parts for each sliding-neighborhood
WD = col2im(sum(X1.*X2,1), [Md Nd], size(F), 'sliding') .* (4/(M*N));
%# pad array with zeros to be of same size as input image
WD = padarray(WD, ([Md Nd]-1)./2, 0, 'both');
end
For what its worth, here is the loop-based version with the improvement that #Laurbert515 suggested:
function WD = wigner_loop(D, Md, Nd, lambda)
%# window size and lambda
if nargin<2, Md = 5; end
if nargin<3, Nd = 5; end
if nargin<4, lambda = 5; end
%# image size
[M,N,~] = size(D);
%# frequency-domain part
F = fft2(D);
WD = zeros([M,N]);
for l = -floor(Nd/2):floor(Nd/2)
for k = -floor(Md/2):floor(Md/2)
%# kernel = exp(-lambda*norm([k,l])
kernel = exp(-lambda * norm([k,l]));
for x = (1+floor(Md/2)):(M-floor(Md/2))
for y = (1+floor(Nd/2)):(N-floor(Nd/2))
%# cos(theta)
theta = 4 * pi * ( real(F(x,y))*k/M + imag(F(x,y))*l/N );
%# f(x+k,y+l) * f(x-k,y-l)* kernel
WD(x,y) = WD(x,y) + ( cos(theta) * D(x+k,y+l) * D(x-k,y-l) * kernel );
end
end
end
end
WD = WD * ( 4/(M*N) );
end
and how I test it (based on what I understood from the paper you previously linked to):
%# difference between two consecutive frames
A = imread('AT3_1m4_02.tif');
B = imread('AT3_1m4_03.tif');
D = imsubtract(A,B);
%#D = rgb2gray(D);
D = im2double(D);
%# apply Wigner-Distribution
tic, WD1 = wigner(D); toc
tic, WD2 = wigner_loop(D); toc
figure(1), imshow(WD1,[])
figure(2), imshow(WD2,[])
you might then need to scale/normalize the matrix, and apply thresholding...
Related
Is there any function in Matlab which calculates the correlation ratio?
Here is an implementation I tried to do, but the results are not right.
function cr = correlation_ratio(X, Y, L)
ni = zeros(1, L);
sigmai = ni;
for i = 0:(L-1)
Yn = Y(X == i);
ni(1, i+1) = numel(Yn);
m = (1/ni(1, i+1))*sum(Yn);
sigmai(1, i+1) = (1/ni(1, i+1))*sum((Yn - m).^2);
end
n = sum(ni);
prod = ni.*sigmai;
cr = (1-(1/n)*sum(prod))^0.5;
This is the equation on the Wikipedia page:
where:
η is the correlation ratio,
yx,i are the sample values (x is the class label, i the sample index),
yx (with the bar on top) is the mean of sample values for class x,
y (with the bar on top) is the mean for all samples across all classes, and
nx is the number of samples in class x.
This is how I interpreted it into code:
function eta = correlation_ratio(X, Y)
X = X(:); % make sure we've got column vectors, simplifies things below a bit
Y = Y(:);
L = max(X);
mYx = zeros(1, L+1); % we'll write mean per class here
nx = zeros(1, L+1); % we'll write number of samples per class here
for i = unique(X).'
Yn = Y(X == i);
if numel(Yn)>1
mYx(i+1) = mean(Yn);
nx(i+1) = numel(Yn);
end
end
mY = mean(Y); % mean across all samples
eta = sqrt(sum(nx .* (mYx - mY).^2) / sum((Y-mY).^2));
The loop could be replaced with accumarray.
I have model with "matlab function block" in which I have recursive least square method. Recursive algorithm needs to know length of incoming signal in order to work correctly. But when I use command N=length(y) it returns me length N= 1. But I think it should give me higher length.
Simulink model
Matlab function block code "rls_iden6"
function [P,N] = fcn(u,y)
%%
N = length(y);
sigma=1;
C = sigma*eye(2); %p
P = ones(2,1);
z= [y; u];
lamda=1;
for n=1:N
sample_out = y(n);
C = (C - ( (C*z*z'*C)/( lamda+(z'*C*z) ) ))/lamda;
P = P + (C*z* (sample_out - (z'*P)));
end
My final code should look like it's shown below, because it works in matlab workspace. Simulink should give me 5 parameters instead of just 2.
load data_cela.mat
u=U; %input
y=Y; %output
%%
input = 3;
output = 2;
system = input + output;
N = length(y);
%initial conditions
sigma = 1;
C = sigma*eye(system);
P = ones(system,1);
lamda = 1; %forgetting factor
for n=3:N
for i=1:2
W(i) = y(n-i); %output
end
for i=1:3
V(i) = u(n-i+1); %input
end
z = [V';W'];
sample_out = y(n);
pom(n)= z' * P;
error(n) = y(n) - pom(n);
C = (C - ( (C*z*z'*C)/( lamda+(z'*C*z) ) ))/lamda;
P = P + (C*z* (sample_out - (z'*P) ) );
change(1:system,n) = P;
end
f_param = [P(1:3);-P(4:5)];
num = [P(1:3,1)];
den = [1;-P(4:5,1)];
num1 = num(3,1);
trasferfunction = tf(num1,den',1)
Result:
0.002879
----------------------
z^2 - 1.883 z + 0.8873
You will need to add a buffer before signal to convert the scalar to matrix. Then after the buffer has been added set the buffer size to the amount of data you want, i.e. by setting it to 2 will make 2 rows and 1 column. This will help you to get the data however, for setting delay properly you will require to set buffer overlap to 1.
Hope this helps.
I'm trying to make a prototype audio recognition system by following this link: http://www.ifp.illinois.edu/~minhdo/teaching/speaker_recognition/. It is quite straightforward so there is almost nothing to worry about. But my problem is with the mel-frequency function. Here is the code as provided on the website:
function m = melfb(p, n, fs)
% MELFB Determine matrix for a mel-spaced filterbank
%
% Inputs: p number of filters in filterbank
% n length of fft
% fs sample rate in Hz
%
% Outputs: x a (sparse) matrix containing the filterbank amplitudes
% size(x) = [p, 1+floor(n/2)]
%
% Usage: For example, to compute the mel-scale spectrum of a
% colum-vector signal s, with length n and sample rate fs:
%
% f = fft(s);
% m = melfb(p, n, fs);
% n2 = 1 + floor(n/2);
% z = m * abs(f(1:n2)).^2;
%
% z would contain p samples of the desired mel-scale spectrum
%
% To plot filterbanks e.g.:
%
% plot(linspace(0, (12500/2), 129), melfb(20, 256, 12500)'),
% title('Mel-spaced filterbank'), xlabel('Frequency (Hz)');
f0 = 700 / fs;
fn2 = floor(n/2);
lr = log(1 + 0.5/f0) / (p+1);
% convert to fft bin numbers with 0 for DC term
bl = n * (f0 * (exp([0 1 p p+1] * lr) - 1));
b1 = floor(bl(1)) + 1;
b2 = ceil(bl(2));
b3 = floor(bl(3));
b4 = min(fn2, ceil(bl(4))) - 1;
pf = log(1 + (b1:b4)/n/f0) / lr;
fp = floor(pf);
pm = pf - fp;
r = [fp(b2:b4) 1+fp(1:b3)];
c = [b2:b4 1:b3] + 1;
v = 2 * [1-pm(b2:b4) pm(1:b3)];
m = sparse(r, c, v, p, 1+fn2);
But it gave me an error:
Error using * Inner matrix dimensions must agree.
Error in MFFC (line 17) z = m * abs(f(1:n2)).^2;
When I include these 2 lines just before line 17:
size(m)
size(abs(f(1:n2)).^2)
It gave me :
ans =
20 65
ans =
1 65
So should I transpose the second matrix? Or should I interpret this as an row-wise multiplication and modify the code?
Edit: Here is the main function (I simply run MFCC()):
function result = MFFC()
[y Fs] = audioread('s1.wav');
% sound(y,Fs)
Frames = Frame_Blocking(y,128);
Windowed = Windowing(Frames);
spectrum = FFT_After_Windowing(Windowed);
%imagesc(mag2db(abs(spectrum)))
p = 20;
S = size(spectrum);
n = S(2);
f = spectrum;
m = melfb(p, n, Fs);
n2 = 1 + floor(n/2);
size(m)
size(abs(f(1:n2)).^2)
z = m * abs(f(1:n2)).^2;
result = z;
And here are the auxiliary functions:
function f = Frame_Blocking(y,N)
% Parameters: M = 100, N = 256
% Default : M = 100; N = 256;
M = fix(N/3);
Frames = [];
first = 1; last = N;
len = length(y);
while last <= len
Frames = [Frames; y(first:last)'];
first = first + M;
last = last + M;
end;
if last < len
first = first + M;
Frames = [Frames; y(first : len)];
end
f = Frames;
function f = Windowing(Frames)
S = size(Frames);
N = S(2);
M = S(1);
Windowed = zeros(M,N);
nn = 1:N;
wn = 0.54 - 0.46*cos(2*pi/(N-1)*(nn-1));
for ii = 1:M
Windowed(ii,:) = Frames(ii,:).*wn;
end;
f = Windowed;
function f = FFT_After_Windowing(Windowed)
spectrum = fft(Windowed);
f = spectrum;
Transpose s or transpose the resulting f (it's just a matter of convention).
There is nothing wrong with the melfb function you are using, merely with the dimensions of the signal in the example you are trying to run (in the commented lines 14-17).
% f = fft(s);
% m = melfb(p, n, fs);
% n2 = 1 + floor(n/2);
% z = m * abs(f(1:n2)).^2;
The example assumes that you are using a "colum-vector signal s". From the size of your Fourier transformed f (done via fft which respects the input signal dimensions) your input signal s is a row-vector signal.
The part that gives you the error is the actual filtering operation that requires multiplying a p x n2 matrix with a n2 x 1 column-vector (i.e., each filter's response is multiplied pointwise with the Fourier of the input signal). Since your input s is 1 x n, your f will be 1 x n and the final matrix to vector multiplication for z will give an error.
Thanks to gevang's anwer, I was able to find out my mistake. Here is how I modified the code:
function result = MFFC()
[y Fs] = audioread('s2.wav');
% sound(y,Fs)
Frames = Frame_Blocking(y,128);
Windowed = Windowing(Frames);
%spectrum = FFT_After_Windowing(Windowed');
%imagesc(mag2db(abs(spectrum)))
p = 20;
%S = size(spectrum);
%n = S(2);
%f = spectrum;
S1 = size(Windowed);
n = S1(2);
n2 = 1 + floor(n/2);
%z = zeros(S1(1),n2);
z = zeros(20,S1(1));
for ii=1: S1(1)
s = (FFT_After_Windowing(Windowed(ii,:)'));
f = fft(s);
m = melfb(p,n,Fs);
% n2 = 1 + floor(n/2);
z(:,ii) = m * abs(f(1:n2)).^2;
end;
%f = FFT_After_Windowing(Windowed');
%S = size(f);
%n = S(2);
%size(f)
%m = melfb(p, n, Fs);
%n2 = 1 + floor(n/2);
%size(m)
%size(abs(f(1:n2)).^2)
%z = m * abs(f(1:n2)).^2;
result = z;
As you can see, I naively assumed that the function deals with row-wise matrices, but in fact it deals with column vectors (and maybe column-wise matrices). So I iterate through each column of the input matrix and then combine the results.
But I don't think this is efficient and vectorized code. Also I still can't figure out how to do column-wise operations on the input matrix (Windowed - after the windowing step), instead of using a loop.
I've written some code to implement an algorithm that takes as input a vector q of real numbers, and returns as an output a complex matrix R. The Matlab code below produces a plot showing the input vector q and the output matrix R.
Given only the complex matrix output R, I would like to obtain the input vector q. Can I do this using least-squares optimization? Since there is a recursive running sum in the code (rs_r and rs_i), the calculation for a column of the output matrix is dependent on the calculation of the previous column.
Perhaps a non-linear optimization can be set up to recompose the input vector q from the output matrix R?
Looking at this in another way, I've used an algorithm to compute a matrix R. I want to run the algorithm "in reverse," to get the input vector q from the output matrix R.
If there is no way to recompose the starting values from the output, thereby treating the problem as a "black box," then perhaps the mathematics of the model itself can be used in the optimization? The program evaluates the following equation:
The Utilde(tau, omega) is the output matrix R. The tau (time) variable comprises the columns of the response matrix R, whereas the omega (frequency) variable comprises the rows of the response matrix R. The integration is performed as a recursive running sum from tau = 0 up to the current tau timestep.
Here are the plots created by the program posted below:
Here is the full program code:
N = 1001;
q = zeros(N, 1); % here is the input
q(1:200) = 55;
q(201:300) = 120;
q(301:400) = 70;
q(401:600) = 40;
q(601:800) = 100;
q(801:1001) = 70;
dt = 0.0042;
fs = 1 / dt;
wSize = 101;
Glim = 20;
ginv = 0;
R = get_response(N, q, dt, wSize, Glim, ginv); % R is output matrix
rows = wSize;
cols = N;
figure; plot(q); title('q value input as vector');
ylim([0 200]); xlim([0 1001])
figure; imagesc(abs(R)); title('Matrix output of algorithm')
colorbar
Here is the function that performs the calculation:
function response = get_response(N, Q, dt, wSize, Glim, ginv)
fs = 1 / dt;
Npad = wSize - 1;
N1 = wSize + Npad;
N2 = floor(N1 / 2 + 1);
f = (fs/2)*linspace(0,1,N2);
omega = 2 * pi .* f';
omegah = 2 * pi * f(end);
sigma2 = exp(-(0.23*Glim + 1.63));
sign = 1;
if(ginv == 1)
sign = -1;
end
ratio = omega ./ omegah;
rs_r = zeros(N2, 1);
rs_i = zeros(N2, 1);
termr = zeros(N2, 1);
termi = zeros(N2, 1);
termr_sub1 = zeros(N2, 1);
termi_sub1 = zeros(N2, 1);
response = zeros(N2, N);
% cycle over cols of matrix
for ti = 1:N
term0 = omega ./ (2 .* Q(ti));
gamma = 1 / (pi * Q(ti));
% calculate for the real part
if(ti == 1)
Lambda = ones(N2, 1);
termr_sub1(1) = 0;
termr_sub1(2:end) = term0(2:end) .* (ratio(2:end).^-gamma);
else
termr(1) = 0;
termr(2:end) = term0(2:end) .* (ratio(2:end).^-gamma);
rs_r = rs_r - dt.*(termr + termr_sub1);
termr_sub1 = termr;
Beta = exp( -1 .* -0.5 .* rs_r );
Lambda = (Beta + sigma2) ./ (Beta.^2 + sigma2); % vector
end
% calculate for the complex part
if(ginv == 1)
termi(1) = 0;
termi(2:end) = (ratio(2:end).^(sign .* gamma) - 1) .* omega(2:end);
else
termi = (ratio.^(sign .* gamma) - 1) .* omega;
end
rs_i = rs_i - dt.*(termi + termi_sub1);
termi_sub1 = termi;
integrand = exp( 1i .* -0.5 .* rs_i );
if(ginv == 1)
response(:,ti) = Lambda .* integrand;
else
response(:,ti) = (1 ./ Lambda) .* integrand;
end
end % ti loop
No, you cannot do so unless you know the "model" itself for this process. If you intend to treat the process as a complete black box, then it is impossible in general, although in any specific instance, anything can happen.
Even if you DO know the underlying process, then it may still not work, as any least squares estimator is dependent on the starting values, so if you do not have a good guess there, it may converge to the wrong set of parameters.
It turns out that by using the mathematics of the model, the input can be estimated. This is not true in general, but for my problem it seems to work. The cumulative integral is eliminated by a partial derivative.
N = 1001;
q = zeros(N, 1);
q(1:200) = 55;
q(201:300) = 120;
q(301:400) = 70;
q(401:600) = 40;
q(601:800) = 100;
q(801:1001) = 70;
dt = 0.0042;
fs = 1 / dt;
wSize = 101;
Glim = 20;
ginv = 0;
R = get_response(N, q, dt, wSize, Glim, ginv);
rows = wSize;
cols = N;
cut_val = 200;
imagLogR = imag(log(R));
Mderiv = zeros(rows, cols-2);
for k = 1:rows
val = deriv_3pt(imagLogR(k,:), dt);
val(val > cut_val) = 0;
Mderiv(k,:) = val(1:end-1);
end
disp('Running iteration');
q0 = 10;
q1 = 500;
NN = cols - 2;
qout = zeros(NN, 1);
for k = 1:NN
data = Mderiv(:,k);
qout(k) = fminbnd(#(q) curve_fit_to_get_q(q, dt, rows, data),q0,q1);
end
figure; plot(q); title('q value input as vector');
ylim([0 200]); xlim([0 1001])
figure;
plot(qout); title('Reconstructed q')
ylim([0 200]); xlim([0 1001])
Here are the supporting functions:
function output = deriv_3pt(x, dt)
% Function to compute dx/dt using the 3pt symmetrical rule
% dt is the timestep
N = length(x);
N0 = N - 1;
output = zeros(N0, 1);
denom = 2 * dt;
for k = 2:N0
output(k - 1) = (x(k+1) - x(k-1)) / denom;
end
function sse = curve_fit_to_get_q(q, dt, rows, data)
fs = 1 / dt;
N2 = rows;
f = (fs/2)*linspace(0,1,N2); % vector for frequency along cols
omega = 2 * pi .* f';
omegah = 2 * pi * f(end);
ratio = omega ./ omegah;
gamma = 1 / (pi * q);
termi = ((ratio.^(gamma)) - 1) .* omega;
Error_Vector = termi - data;
sse = sum(Error_Vector.^2);
I'm trying to implement a very basic eigenface calculation in Matlab. It kind of works but I get only two meaningful eigenvalues - the rest are zero. The corresponding eigenvectors seem to be right since most of them will show an eigenface when converting to an image.
So why are most of my eigenvalues zero? I need them to be different from zero in order to sort the eigenfaces by their significance (greatest magnitude eigenvalues).
I am reading 400 images, each size h/w = 112/92 px
They can be found here: http://www.cl.cam.ac.uk/Research/DTG/attarchive/pub/data/att_faces.zip
The code:
clear all;
files = dir('eigenfaces2/training/*.pgm');
[numFaces, discard] = size(files);
h = 112;
w = 92;
s = h * w;
%calculate average face
avgFace = zeros(s, 1);
faces = [];
for i=1:numFaces
file = strcat('eigenfaces2/training/', files(i).name);
im = double(imread(file));
im = reshape(im, s, 1);
avgFace = avgFace + im;
faces(:,i) = im;
end
avgFace = avgFace ./ numFaces;
A = [];
for i=1:numFaces
diff = avgFace - faces(i);
A(:,i) = diff;
end
numEigs = 20;
L = (A' * A) / numFaces;
[tmpEigs, discard] = eigs(L, numEigs);
eigenfaces = [];
for i=1:numEigs
v = tmpEigs(:,i);
eigenfaces(:,i) = A * v;
end
%visualize largest eigenfaces
figure;
for i=1:numEigs
eigface = eigenfaces(:,i);
mmax = max(eigface);
mmin = min(eigface);
eigface = 255 .* (eigface-mmin) ./ (mmax-mmin);
eigface = reshape(eigface, h, w);
subplot(4,5,i); imshow(uint8(eigface));
end
I've don't have much experience with computer vision/image recognition, but I think you might want
diff = avgFace - faces(:,i);
in your second for loop. Otherwise it's just subtracting a constant from avgFace each time, and so A (and hence L) only gets a rank of 2.