I have the output 2D array C(kx,ky) with Fourier coefficients and the problem is in how to get the function f(x,y) back using that coefficients.
f(x,y) is defined on [0,2Pi] x [0,2Pi] space in Fourier basis with 64 points in each direction. My output array with coefficients has the size (32,63). 32 represents only coefficients for positive wave numbers k_x > 0 and 63 is for -k_y < 0 < k_y.
In order to get the coefficients for negative k_x < 0 I do conjugate symmetry operation for C(kx,ky) i.e. np.conjugate(C) in python.
Getting back f(x,y):
#1st part from for k_x => 0:
for i in range(len(kx)):
for j in range(len(ky)):
summode = summode + c[i,j]*np.exp(1j*(kx[i]*x + ky[j]*y))
#2nd part for k_x < 0:
for i in range(1,len(kx)):
for j in range(len(ky)):
summode = summode + np.conjugate(c[i,j])*np.exp(1j*(-kx[i]*x + ky[j]*y))
In the end, my obtained function does not match the original. Any ideas?
Thanks in advance!
Related
There is an unknown function ,
and there are unknown coefficients k, l. Task is to estimate k, l using linear regression, through the data table.
-2.0 1.719334581463762
-1.0 1.900158577875515
0.0 2.1
1.0 2.3208589279588603
2.0 2.5649457921363568
Till now mathematically I did like, taking logarithm on both sides
Then using the data table, 5 equations will be formed
Now apply the linear regressor, to this logarithm-transformed data, to estimate the coefficients k and l.
I have built a linear regresor,
using DataFrames, GLM
function LinearRegression(X)
x = X[:,1]
y = X[:,2]
data = DataFrame(y = y, x = x)
reg = lm(#formula(y ~ x), data)
return coef(reg)[2], coef(reg)[1]
end
Any solution to how to find l and k values using this technique?
You're almost there, but I think you have a misconception mathematically in your code. You are right that taking the log of f(x) makes this essentially a linear fit (of form y = mx + b) but you haven't told the code that, i.e. your LinearRegression function should read:
function LinearRegression(X)
x = X[:,1]
y = X[:,2]
data = DataFrame(y = log.(y), x = x)
reg = lm(#formula(y ~ x), data)
return coef(reg)[2], coef(reg)[1]
end
Note that I have written y = log.(y) to match the formula as otherwise you are fitting a line to exponential data. We don't take the log of x because it has negative values. Your function will then return the correct coefficients l and log(k) (so if you want just k itself you need to take the exponential) -- see this plot as proof that it fits the data perfectly!
You need to convert the intercept with exp and the slope keeps as it is.
using Statistics #mean
#Data
X = [-2.0 1.719334581463762
-1.0 1.900158577875515
0.0 2.1
1.0 2.3208589279588603
2.0 2.5649457921363568]
x = X[:,1]
y = X[:,2]
yl = log.(y)
#Get a and b for: log.(y) = a + b*x
b = x \ yl
a = mean(yl) - b * mean(x)
l = b
#0.10000000000000005
k = exp(a)
#2.1
k*exp.(l.*x)
#5-element Vector{Float64}:
# 1.719334581463762
# 1.900158577875515
# 2.1
# 2.3208589279588603
# 2.5649457921363568
I would like to smooth an Impulse Response audio file. The FFT of the file shows that it is very spikey. I would like to smooth out the audio file, not just its plot, so that I have a smoother IR file.
I have found a function that shows the FFT plot smoothed out. How could this smoothing be applied to the actual FFT data and not just to the plot of it?
[y,Fs] = audioread('test\test IR.wav');
function x_oct = smoothSpectrum(X,f,Noct)
%SMOOTHSPECTRUM Apply 1/N-octave smoothing to a frequency spectrum
%% Input checking
assert(isvector(X), 'smoothSpectrum:invalidX', 'X must be a vector.');
assert(isvector(f), 'smoothSpectrum:invalidF', 'F must be a vector.');
assert(isscalar(Noct), 'smoothSpectrum:invalidNoct', 'NOCT must be a scalar.');
assert(isreal(X), 'smoothSpectrum:invalidX', 'X must be real.');
assert(all(f>=0), 'smoothSpectrum:invalidF', 'F must contain positive values.');
assert(Noct>=0, 'smoothSpectrum:invalidNoct', 'NOCT must be greater than or equal to 0.');
assert(isequal(size(X),size(f)), 'smoothSpectrum:invalidInput', 'X and F must be the same size.');
%% Smoothing
% calculates a Gaussian function for each frequency, deriving a
% bandwidth for that frequency
x_oct = X; % initial spectrum
if Noct > 0 % don't bother if no smoothing
for i = find(f>0,1,'first'):length(f)
g = gauss_f(f,f(i),Noct);
x_oct(i) = sum(g.*X); % calculate smoothed spectral coefficient
end
% remove undershoot when X is positive
if all(X>=0)
x_oct(x_oct<0) = 0;
end
end
endfunction
function g = gauss_f(f_x,F,Noct)
% GAUSS_F calculate frequency-domain Gaussian with unity gain
%
% G = GAUSS_F(F_X,F,NOCT) calculates a frequency-domain Gaussian function
% for frequencies F_X, with centre frequency F and bandwidth F/NOCT.
sigma = (F/Noct)/pi; % standard deviation
g = exp(-(((f_x-F).^2)./(2.*(sigma^2)))); % Gaussian
g = g./sum(g); % normalise magnitude
endfunction
% take fft
Y = fft(y);
% keep only meaningful frequencies
NFFT = length(y);
if mod(NFFT,2)==0
Nout = (NFFT/2)+1;
else
Nout = (NFFT+1)/2;
end
Y = Y(1:Nout);
f = ((0:Nout-1)'./NFFT).*Fs;
% put into dB
Y = 20*log10(abs(Y)./NFFT);
% smooth
Noct = 12;
Z = smoothSpectrum(Y,f,Noct);
% plot
semilogx(f,Y,'LineWidth',0.7,f,Z,'LineWidth',2.2);
xlim([20,20000])
grid on
PS. I have Octave GNU, so I don't have the functions that are available with Matlab Toolboxes.
Here is the test IR audio file.
I think I found it. Since the FFT of the audio file (which is real numbers) is symmetric, with the same real part on both sides but opposite imaginary part, I thought of doing this:
take the FFT, keep the half of it, and apply the smoothing function without converting the magnitudes to dB
then make a copy of that smoothed FFT, and invert just the imaginary part
combine the two parts so that I have the same symmetric FFT as I had in the beginning, but now it is smoothed
apply inverse FFT to this and take the real part and write it to file.
Here is the code:
[y,Fs] = audioread('test IR.wav');
function x_oct = smoothSpectrum(X,f,Noct)
x_oct = X; % initial spectrum
if Noct > 0 % don't bother if no smoothing
for i = find(f>0,1,'first'):length(f)
g = gauss_f(f,f(i),Noct);
x_oct(i) = sum(g.*X); % calculate smoothed spectral coefficient
end
% remove undershoot when X is positive
if all(X>=0)
x_oct(x_oct<0) = 0;
end
end
endfunction
function g = gauss_f(f_x,F,Noct)
sigma = (F/Noct)/pi; % standard deviation
g = exp(-(((f_x-F).^2)./(2.*(sigma^2)))); % Gaussian
g = g./sum(g); % normalise magnitude
endfunction
% take fft
Y = fft(y);
% keep only meaningful frequencies
NFFT = length(y);
if mod(NFFT,2)==0
Nout = (NFFT/2)+1;
else
Nout = (NFFT+1)/2;
end
Y = Y(1:Nout);
f = ((0:Nout-1)'./NFFT).*Fs;
% smooth
Noct = 12;
Z = smoothSpectrum(Y,f,Noct);
% plot
semilogx(f,Y,'LineWidth',0.7,f,Z,'LineWidth',2.2);
xlim([20,20000])
grid on
#Apply the smoothing to the actual data
Zreal = real(Z); # real part
Zimag_neg = Zreal - Z; # opposite of imaginary part
Zneg = Zreal + Zimag_neg; # will be used for the symmetric Z
# Z + its symmetry with same real part but opposite imaginary part
reconstructed = [Z ; Zneg(end-1:-1:2)];
# Take the real part of the inverse FFT
reconstructed = real(ifft(reconstructed));
#Write to file
audiowrite ('smoothIR.wav', reconstructed, Fs, 'BitsPerSample', 24);
Seems to work! :) It would be nice if someone more knowledgeable could confirm that the thinking and code are good :)
I'm attempting my first solo project, after taking an introductory course to machine learning, where I'm trying to use linear approximation to predict the outcome of addition/subtraction of two numbers.
I have 3 features: first number, subtraction/addition (0 or 1), and second number.
So my input looks something like this:
3 0 1
4 1 2
3 0 3
With corresponding output like this:
2
6
0
I have (I think) successfully implemented logistic regression algorithm, as the squared error does gradually decrease, but in 100 values, ranging from 0 to 50, the squared error value flattens out at around 685.6 after about 400 iterations.
Graph: Squared Error vs Iterations
.
To fix this, I have tried using a larger dataset for training, getting rid of regularization, and normalizing the input values.
I know that one of the steps to fix high bias is to add complexity to the approximation, but I want to maximize the performance at this particular level. Is it possible to go any further on this level?
My linear approximation code in Octave:
% Iterate
for i = 1 : iter
% hypothesis
h = X * Theta;
% reg theta prep
regTheta = Theta;
regTheta(:, 1) = 0;
% cost calc
J(i, 2) = (1 / (2 * m)) * (sum((h - y) .^ 2) + lambda * sum(sum(regTheta .^ 2,1),2));
% theta calc
Theta = Theta - (alpha / m) * ((h - y)' * X)' + lambda * sum(sum(regTheta, 1), 2);
end
Note: I'm using 0 for lambda, as to ignore regularization.
function [theta, J_history] = gradientDescent(X, y, theta, alpha, num_iters)
%GRADIENTDESCENT Performs gradient descent to learn theta
% theta = GRADIENTDESENT(X, y, theta, alpha, num_iters) updates theta by
% taking num_iters gradient steps with learning rate alpha
% Initialize some useful values
m = length(y); % number of training examples
J_history = zeros(num_iters, 1);
for iter = 1:num_iters
% ====================== YOUR CODE HERE ======================
% Instructions: Perform a single gradient step on the parameter vector
% theta.
%
% Hint: While debugging, it can be useful to print out the values
% of the cost function (computeCost) and gradient here.
%
hypothesis = x*theta;
theta_0 = theta(1) - alpha(1/m)*sum((hypothesis-y)*x);
theta_1 = theta(2) - alpha(1/m)*sum((hypothesis-y)*x);
theta(1) = theta_0;
theta(2) = theta_1;
% ============================================================
% Save the cost J in every iteration
J_history(iter) = computeCost(X, y, theta);
end
end
I keep getting this error
error: gradientDescent: subscript indices must be either positive integers less than 2^31 or logicals
on this line right in-between the first theta and =
theta_0 = theta(1) - alpha(1/m)*sum((hypothesis-y)*x);
I'm very new to octave so please go easy on me, and
thank you in advance.
This is from the coursera course on Machine Learning from Week 2
99% sure your error is on the line pointed out by topsig, where you have alpha(1/m)
it would help if you gave an example of input values to your function and what you hoped to see as an output, but I'm assuming from your comment
% taking num_iters gradient steps with learning rate alpha
that alpha is a constant, not a function. as such, you have the line alpha(1/m) without any operator in between. octave sees this as you indexing alpha with the value of 1/m.
i.e., if you had an array
x = [3 4 5]
x*(2) = [6 8 10] %% two times each element in the array
x(2) = [4] %% second element in the array
what you did doesn't seem to make sense, as 'm = length(y)' which will output a scalar, so
x = [3 4 5]; m = 3;
x*(1/m) = x*(1/3) = [1 1.3333 1.6666] %% element / 3
x(1/m) = ___error___ %% the 1/3 element in the array makes no sense
note that for certain errors it always indicates that the location of the error is at the assignment operator (the equal sign at the start of the line). if it points there, you usually have to look elsewhere in the line for the actual error. here, it was yelling at you for trying to apply a non-integer subscript (1/m)
If I have an FFT implementation of a certain size M (power of 2), how can I calculate the FFT of a set of size P=k*M, where k is a power of 2 as well?
#define M 256
#define P 1024
complex float x[P];
complex float X[P];
// Use FFT_M(y) to calculate X = FFT_P(x) here
[The question is expressed in a general sense on purpose. I know FFT calculation is a huge field and many architecture specific optimizations were researched and developed, but what I am trying to understand is how is this doable in the more abstract level. Note that I am no FFT (or DFT, for that matter) expert, so if an explanation can be laid down in simple terms that would be appreciated]
Here's an algorithm for computing an FFT of size P using two smaller FFT functions, of sizes M and N (the original question call the sizes M and k).
Inputs:
P is the size of the large FFT you wish to compute.
M, N are selected such that MN=P.
x[0...P-1] is the input data.
Setup:
U is a 2D array with M rows and N columns.
y is a vector of length P, which will hold FFT of x.
Algorithm:
step 1. Fill U from x by columns, so that U looks like this:
x(0) x(M) ... x(P-M)
x(1) x(M+1) ... x(P-M+1)
x(2) x(M+2) ... x(P-M+2)
... ... ... ...
x(M-1) x(2M-1) ... x(P-1)
step 2. Replace each row of U with its own FFT (of length N).
step 3. Multiply each element of U(m,n) by exp(-2*pi*j*m*n/P).
step 4. Replace each column of U with its own FFT (of length M).
step 5. Read out the elements of U by rows into y, like this:
y(0) y(1) ... y(N-1)
y(N) y(N+1) ... y(2N-1)
y(2N) y(2N+1) ... y(3N-1)
... ... ... ...
y(P-N) y(P-N-1) ... y(P-1)
Here is MATLAB code which implements this algorithm. You can test it by typing fft_decomposition(randn(256,1), 8);
function y = fft_decomposition(x, M)
% y = fft_decomposition(x, M)
% Computes FFT by decomposing into smaller FFTs.
%
% Inputs:
% x is a 1D array of the input data.
% M is the size of one of the FFTs to use.
%
% Outputs:
% y is the FFT of x. It has been computed using FFTs of size M and
% length(x)/M.
%
% Note that this implementation doesn't explicitly use the 2D array U; it
% works on samples of x in-place.
q = 1; % Offset because MATLAB starts at one. Set to 0 for C code.
x_original = x;
P = length(x);
if mod(P,M)~=0, error('Invalid block size.'); end;
N = P/M;
% step 2: FFT-N on rows of U.
for m = 0 : M-1
x(q+(m:M:P-1)) = fft(x(q+(m:M:P-1)));
end;
% step 3: Twiddle factors.
for m = 0 : M-1
for n = 0 : N-1
x(m+n*M+q) = x(m+n*M+q) * exp(-2*pi*j*m*n/P);
end;
end;
% step 4: FFT-M on columns of U.
for n = 0 : N-1
x(q+n*M+(0:M-1)) = fft(x(q+n*M+(0:M-1)));
end;
% step 5: Re-arrange samples for output.
y = zeros(size(x));
for m = 0 : M-1
for n = 0 : N-1
y(m*N+n+q) = x(m+n*M+q);
end;
end;
err = max(abs(y-fft(x_original)));
fprintf( 1, 'The largest error amplitude is %g\n', err);
return;
% End of fft_decomposition().
kevin_o's response worked quite well. I took his code and eliminated the loops using some basic Matlab tricks. It functionally is identical to his version
function y = fft_decomposition(x, M)
% y = fft_decomposition(x, M)
% Computes FFT by decomposing into smaller FFTs.
%
% Inputs:
% x is a 1D array of the input data.
% M is the size of one of the FFTs to use.
%
% Outputs:
% y is the FFT of x. It has been computed using FFTs of size M and
% length(x)/M.
%
% Note that this implementation doesn't explicitly use the 2D array U; it
% works on samples of x in-place.
q = 1; % Offset because MATLAB starts at one. Set to 0 for C code.
x_original = x;
P = length(x);
if mod(P,M)~=0, error('Invalid block size.'); end;
N = P/M;
% step 2: FFT-N on rows of U.
X=fft(reshape(x,M,N),[],2);
% step 3: Twiddle factors.
X=X.*exp(-j*2*pi*(0:M-1)'*(0:N-1)/P);
% step 4: FFT-M on columns of U.
X=fft(X);
% step 5: Re-arrange samples for output.
x_twiddle=bsxfun(#plus,M*(0:N-1)',(0:M-1))+q;
y=X(x_twiddle(:));
% err = max(abs(y-fft(x_original)));
% fprintf( 1, 'The largest error amplitude is %g\n', err);
return;
% End of fft_decomposition()
You could just use the last log2(k) passes of a radix-2 FFT, assuming the previous FFT results are from appropriately interleaved data subsets.
Well an FFT is basically a recursive type of Fourier Transform. It relies on the fact that as wikipedia puts it:
The best-known FFT algorithms depend upon the factorization of N, but there are FFTs with O(N log N) complexity for >all N, even for prime N. Many FFT algorithms only depend on the fact that e^(-2pi*i/N) is an N-th primitive root of unity, and >thus can be applied to analogous transforms over any finite field, such as number-theoretic transforms. Since the >inverse DFT is the same as the DFT, but with the opposite sign in the exponent and a 1/N factor, any FFT algorithm >can easily be adapted for it.
So this has pretty much already been done in the FFT. If you are talking about getting longer period signals out of your transform you are better off doing an DFT over the data sets of limited frequencies. There might be a way to do it from the frequency domain but IDK if anyone has actually done it. You could be the first!!!! :)