Is there an easy way to calculate the frequency response of the following function?
I tried using heaviside function but with no luck.
Basically I want to write a function to return the frequency response based on input N1 and N2 and also the number of points (lets say x) between 0 and pi
The output would be a vector which returns x values for the frequency response for corresponding frequencies => 0:pi/x:pi
Assuming that N1 + N2 < num_points, where num_points is the length of the sequence, you can simply write the function like so:
function [gr] = rosenburg(N1, N2, num_points)
gr = zeros(num_points,1);
range1 = 0:N1;
range2 = N1+1:N1+N2;
gr(range1+1) = 0.5*(1 - cos(pi*range1/N1));
gr(range2+1) = cos(pi*(range2-N1) / (2*N2));
end
The function prototype, rosenburg takes in N1, N2 and the total number of points you want this function to take in, num_points. How this code works is that we first allocate an array that is all zeroes of size num_points. We then compute two linear ranges: One from 0 <= n <= N1 and the other from N1 < n <= N2. Note that the second range starts by offsetting N1 by 1 because we have already computed the value at n = N1. Once we compute these ranges, we simply apply the right relationship in the right ranges. Note that when I'm assigning the relationships to the correct intervals in the array, I need to offset by 1 because MATLAB begins indexing arrays at index 1. The rest of the values are zero due to the initialization at the beginning of the function.
Now, if you want to find the frequency response of this signal, just use fft which is the Fast Fourier Transform. It's the classic method to find the frequency domain version of a discrete input signal on a numerical basis. As such, once you create your signal using the rosenburg function, then throw this into the FFT function. How you call it is like so:
X = fft(gr);
This computes the N point FFT, where N is the length of the signal gr. Alternatively, you can provide the number of points you want to compute the FFT for. Specifically:
X = fft(gr, N);
Basically, the higher N is, the finer or granular the frequency components will be. Note that the frequency axis is normalized between 0 to 2*pi, and so the higher N is, the finer resolution you will have between neighbouring points on the axis. Specifically, each point on this axis has the following frequency:
w = i*(2*pi)/x;
i would be the index on the x-axis (0, 1, 2, ..., num_points-1) and x would be the total number of points for the FFT. Normally, people show the spectrum between -pi <= w <= pi, and so some people apply fftshift to shift the spectrum so that the DC component is located at the centre of the spectrum, which is how we naturally perceive the spectrum to be.
When you say "frequency response", I believe you are referring to the magnitude, and so use abs to calculate the complex magnitude of each value, as the fft is generally complex valued. Therefore, assuming that you wish to compute the FFT to be as many points as the length of your signal, and let's say we choose N1 = 4, N2 = 8 and we want 64 points, and we want to plot the spectrum. Simply do this:
gr = rosenburg(4, 8, 64);
X = fft(gr);
Xshift = fftshift(X);
plot(linspace(-pi,pi,64), abs(Xshift));
grid;
The above code will shift the spectrum, then plot its magnitude between -pi to pi. This is what I get:
As an illustration, this is what the spectrum looks like before we apply fftshift:
Here's the code to generate the above figure:
plot(linspace(0,2*pi,64), abs(X));
grid;
You can see that the spectra is symmetric. Right at the frequency pi, you can see that it is mirror reflected, which makes sense as the range from pi to 2*pi, precisely maps to -pi to 0. Because the signal is real, the spectrum is symmetric. In fact, we can call this signal Hermitian symmetric. Obviously, the frequency components are a bit sparsely spaced. It may be better to increase the total number of points to something like 256. This is what I get when I change the number of points to 256:
Pretty smooth! Now, if you want to extract the frequency components from 0 to pi, you need to extract half of the frequency decomposition that is stored in X. Therefore, you would simply do:
f = X(1:numel(X)/2);
numel determines how many elements are in an array or matrix. However, remember that each frequency point was defined as:
w = i*(2*pi)/x
You specifically want:
w = i*pi/x
As such, you'll need to compute the FFT at twice the size of your signal first, then extract half of the spectra in the same way. For example, for 64 points:
gr = rosenburg(4, 8, 64);
X = fft(gr, 128);
f = X(1:numel(X)/2);
This should hopefully get you started. Good luck!
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 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 am trying to compare the FFT of exp(-t^2) to the function's analytical fourier transform, exp(-(w^2)/4)/sqrt(2), over the frequency range -3 to 3.
I have written the following matlab code and have iterated on it MANY times now with no success.
fs = 100; %sampling frequency
dt = 1/fs;
t = 0:dt:10-dt; %time vector
L = length(t); %number of sample points
%N = 2^nextpow2(L); %necessary?
y = exp(-(t.^2));
Y=dt*ifftshift(abs(fft(y)));
freq = (-L/2:L/2-1)*fs/L; %freq vector
F = (exp(-(freq.^2)/4))/sqrt(2); %analytical solution
%Y_valid_pts = Y(W>=-3 & W<=3); %compare for freq = -3 to 3
%npts = length(Y_valid_pts);
% w = linspace(-3,3,npts);
% Fe = (exp(-(w.^2)/4))/sqrt(2);
error = norm(Y - F) %L2 Norm for error
hold on;
plot(freq,Y,'r');
plot(freq,F,'b');
xlabel('Frequency, w');
legend('numerical','analytic');
hold off;
You can see that right now, I am simply trying to get the two plots to look similar. Eventually, I would like to find a way to do two things:
1) find the minimum sampling rate,
2) find the minimum number of samples,
to reach an error (defined as the L2 norm of the difference between the two solutions) of 10^-4.
I feel that this is pretty simple, but I can't seem to even get the two graphs visually agree.
If someone could let me know where I'm going wrong and how I can tackle the two points above (minimum sampling frequency and minimum number of samples) I would be very appreciative.
Thanks
A first thing to note is that the Fourier transform pair for the function exp(-t^2) over the +/- infinity range, as can be derived from tables of Fourier transforms is actually:
Finally, as you are generating the function exp(-t^2), you are limiting the range of t to positive values (instead of taking the whole +/- infinity range).
For the relationship to hold, you would thus have to generate exp(-t^2) with something such as:
t = 0:dt:10-dt; %time vector
t = t - 0.5*max(t); %center around t=0
y = exp(-(t.^2));
Then, the variable w represents angular frequency in radians which is related to the normalized frequency freq through:
w = 2*pi*freq;
Thus,
F = (exp(-((2*pi*freq).^2)/4))*sqrt(pi); %analytical solution
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 a vector of data, which contains integers in the range -20 20.
Bellow is a plot with the values:
This is a sample of 96 elements from the vector data. The majority of the elements are situated in the interval -2, 2, as can be seen from the above plot.
I want to eliminate the noise from the data. I want to eliminate the low amplitude peaks, and keep the high amplitude peak, namely, peaks like the one at index 74.
Basically, I just want to increase the contrast between the high amplitude peaks and low amplitude peaks, and if it would be possible to eliminate the low amplitude peaks.
Could you please suggest me a way of doing this?
I have tried mapstd function, but the problem is that it also normalizes that high amplitude peak.
I was thinking at using the wavelet transform toolbox, but I don't know exact how to reconstruct the data from the wavelet decomposition coefficients.
Can you recommend me a way of doing this?
One approach to detect outliers is to use the three standard deviation rule. An example:
%# some random data resembling yours
x = randn(100,1);
x(75) = -14;
subplot(211), plot(x)
%# tone down the noisy points
mu = mean(x); sd = std(x); Z = 3;
idx = ( abs(x-mu) > Z*sd ); %# outliers
x(idx) = Z*sd .* sign(x(idx)); %# cap values at 3*STD(X)
subplot(212), plot(x)
EDIT:
It seems I misunderstood the goal here. If you want to do the opposite, maybe something like this instead:
%# some random data resembling yours
x = randn(100,1);
x(75) = -14; x(25) = 20;
subplot(211), plot(x)
%# zero out everything but the high peaks
mu = mean(x); sd = std(x); Z = 3;
x( abs(x-mu) < Z*sd ) = 0;
subplot(212), plot(x)
If it's for demonstrative purposes only, and you're not actually going to be using these scaled values for anything, I sometimes like to increase contrast in the following way:
% your data is in variable 'a'
plot(a.*abs(a)/max(abs(a)))
edit: since we're posting images, here's mine (before/after):
You might try a split window filter. If x is your current sample, the filter would look something like:
k = [L L L L L L 0 0 0 x 0 0 0 R R R R R R]
For each sample x, you average a band of surrounding samples on the left (L) and a band of surrounding samples on the right. If your samples are positive and negative (as yours are) you should take the abs. value first. You then divide the sample x by the average value of these surrounding samples.
y[n] = x[n] / mean(abs(x([L R])))
Each time you do this the peaks are accentuated and the noise is flattened. You can do more than one pass to increase the effect. It is somewhat sensitive to the selection of the widths of these bands, but can work. For example:
Two passes:
What you actually need is some kind of compression to scale your data, that is: values between -2 and 2 are scale by a certain factor and everything else is scaled by another factor. A crude way to accomplish such a thing, is by putting all small values to zero, i.e.
x = randn(1,100)/2; x(50) = 20; x(25) = -15; % just generating some data
threshold = 2;
smallValues = (abs(x) <= threshold);
y = x;
y(smallValues) = 0;
figure;
plot(x,'DisplayName','x'); hold on;
plot(y,'r','DisplayName','y');
legend show;
Please do not that this is a very nonlinear operation (e.g. when you have wanted peaks valued at 2.1 and 1.9, they will produce very different behavior: one will be removed, the other will be kept). So for displaying, this might be all you need, for further processing it might depend on what you are trying to do.
To eliminate the low amplitude peaks, you're going to equate all the low amplitude signal to noise and ignore.
If you have any apriori knowledge, just use it.
if your signal is a, then
a(abs(a)<X) = 0
where X is the max expected size of your noise.
If you want to get fancy, and find this "on the fly" then, use kmeans of 3. It's in the statistics toolbox, here:
http://www.mathworks.com/help/toolbox/stats/kmeans.html
Alternatively, you can use Otsu's method on the absolute values of the data, and use the sign back.
Note, these and every other technique I've seen on this thread is assuming you are doing post processing. If you are doing this processing in real time, things will have to change.