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I have the following image:
The coordinates corresponding to the white blobs in the image are sorted according to the increasing value of x-coordinate. However, I want them to follow the following pattern:
(In a zig-zag manner from bottom left to top left.)
Any clue how can I go about it? Any clue regarding the algorithm will be appreciated.
The set of coordinates are as follows:
[46.5000000000000,104.500000000000]
[57.5000000000000,164.500000000000]
[59.5000000000000,280.500000000000]
[96.5000000000000,66.5000000000000]
[127.500000000000,103.500000000000]
[142.500000000000,34.5000000000000]
[156.500000000000,173.500000000000]
[168.500000000000,68.5000000000000]
[175.500000000000,12.5000000000000]
[198.500000000000,37.5000000000000]
[206.500000000000,103.500000000000]
[216.500000000000,267.500000000000]
[225.500000000000,14.5000000000000]
[234.500000000000,62.5000000000000]
[251.500000000000,166.500000000000]
[258.500000000000,32.5000000000000]
[271.500000000000,13.5000000000000]
[284.500000000000,103.500000000000]
[291.500000000000,61.5000000000000]
[313.500000000000,32.5000000000000]
[318.500000000000,10.5000000000000]
[320.500000000000,267.500000000000]
[352.500000000000,57.5000000000000]
[359.500000000000,102.500000000000]
[360.500000000000,167.500000000000]
[366.500000000000,11.5000000000000]
[366.500000000000,34.5000000000000]
[408.500000000000,9.50000000000000]
[414.500000000000,62.5000000000000]
[419.500000000000,34.5000000000000]
[451.500000000000,12.5000000000000]
[456.500000000000,97.5000000000000]
[457.500000000000,168.500000000000]
[465.500000000000,62.5000000000000]
[465.500000000000,271.500000000000]
[468.500000000000,31.5000000000000]
[498.500000000000,10.5000000000000]
[522.500000000000,105.500000000000]
[524.500000000000,32.5000000000000]
[533.500000000000,60.5000000000000]
[534.500000000000,11.5000000000000]
[565.500000000000,164.500000000000]
[576.500000000000,33.5000000000000]
[581.500000000000,10.5000000000000]
[582.500000000000,67.5000000000000]
[586.500000000000,267.500000000000]
[590.500000000000,102.500000000000]
[622.500000000000,10.5000000000000]
[630.500000000000,32.5000000000000]
[646.500000000000,58.5000000000000]
[653.500000000000,94.5000000000000]
[669.500000000000,8.50000000000000]
[678.500000000000,167.500000000000]
[680.500000000000,31.5000000000000]
[705.500000000000,57.5000000000000]
[719.500000000000,9.50000000000000]
[729.500000000000,271.500000000000]
[732.500000000000,33.5000000000000]
[733.500000000000,97.5000000000000]
[757.500000000000,11.5000000000000]
[758.500000000000,59.5000000000000]
[778.500000000000,157.500000000000]
[792.500000000000,31.5000000000000]
[802.500000000000,10.5000000000000]
[812.500000000000,94.5000000000000]
[834.500000000000,59.5000000000000]
[839.500000000000,30.5000000000000]
[865.500000000000,160.500000000000]
[866.500000000000,272.500000000000]
[885.500000000000,58.5000000000000]
[892.500000000000,97.5000000000000]
[955.500000000000,94.5000000000000]
[963.500000000000,163.500000000000]
[972.500000000000,265.500000000000]
Building upon uSeemSurprised's answer, I would go for a 3-steps approach:
Sort the points list by y-coord. This is O(n log n)
Determine the y-axis ranges. I simply iterate over the points and take note of where the y-coord difference is larger than a threshold value. This is O(n) of course
Sort each of the sublists that represent the y-axis lines by x-coord. If we had m sublists of k items each this would be O(m (k log k)); so the overall process is still O(n log n)
The code:
def zigzag(points, threshold=10.0)
#step 1
points.sort(key=lambda x:x[1])
#step 2
breaks = []
for i in range(1, len(points)):
if points[i][1] - points[i-1][1] > threshold:
breaks.append(i)
breaks.append(i)
#step 3
rev = False
start = 0
outpoints = []
for b in breaks:
outpoints += sorted(points[start:b], reverse = rev)
start = b
rev = not rev
return outpoints
You can sort the x-axis coordinates corresponding to y-axis coordinates, where you consider certain y-axis range, i.e the coordinates that are sorted according to x-axis all belong to the same y-axis range. Each time you move up to a different y-axis range you can flip the sorting order, i.e increasing then decreasing and so on.
The most similar algorithm I can think of is Andrew's algorithm for convex hulls, specifically the lower hull (though depending on the coordinate system, you may need to use the upper hull instead).
Running the lower hull algorithm and removing points until no points remain would get you want. To get the zig-zag patterning, reverse the ordering every other time you run it.
Here is implementations in most languages:
https://en.wikibooks.org/wiki/Algorithm_Implementation/Geometry/Convex_hull/Monotone_chain
Edit: Downside here is precision in the case of fuzzy measurements. You may need to adjust the algorithm a bit if convex hulls aren't exactly what you need. IE: if you want to consider it still part of the hull if it's within say with 0.1 or say 1% of being on the hull or something. In the example given, the coordinates are exactly on the line so it would work well, but not so much so if the coordinates were say randomly distributed within say 0.1 of their actual positions.
This approach assumes you know how many rows you expect, although I suspect there's programmatic ways you could estimate that.
nbins = 6; % Number of horizontal rows we expect
[bin,binC] = kmedoids(A(:,2),nbins); % Use a clustering approach to group them
bin = binC(bin); % Clusters in random order, fix it so that clusters
[~,~,bin] = unique(bin); % are ordered by central y value
xord = A(:,1) .* (-1).^mod(bin+1,2); % flip/flop for each row making the x-coord +ve or -ve
% so that we can sort in a zig-zag
[~,idx] = sortrows([bin,xord], [1,2]); % Sort by the clusters and the zig-zag
B = A( idx, : ); % Create re-ordered array
Plotting this, it seems like what you want
figure(99); clf; hold on;
plot( A(:,1), A(:,2), '-o' );
plot( B(:,1), B(:,2), '-', 'linewidth', 1.5 );
set(gca, 'YDir', 'reverse');
legend( {'Original','Reordered'} );
Use a nearest neighbor search, where you define a custom distance measure which makes distance in the Y direction more expensive than distance in the X direction. Then start the algorithm with the bottom left point.
The "normal" Euclidean distance in Cartesian coordinates is calculated by sqrt( (x2 - x1)^2 + (y2 - y1)^2 )
To make the y direction more expensive, use a custom distance formula where you multiply the y result by a constant:
sqrt( (x2 - x1)^2 + k*(y2 - y1)^2 )
where the constant k is larger than 1 but not much larger, I would start with 2.
currently I'm needing a function which gives a weighted, random number.
It should chose a random number between two doubles/integers (for example 4 and 8) while the value in the middle (6) will occur on average, about twice as often than the limiter values 4 and 8.
If this were only about integers, I could predefine the values with variables and custom probabilities, but I need the function to give a double with at least 2 digits (meaning thousands of different numbers)!
The environment I use, is the "Game Maker" which provides all sorts of basic random-generators, but not weighted ones.
Could anyone possibly lead my in the right direction how to achieve this?
Thanks in advance!
The sum of two independent continuous uniform(0,1)'s, U1 and U2, has a continuous symmetrical triangle distribution between 0 and 2. The distribution has its peak at 1 and tapers to zero at either end. We can easily translate that to a range of (4,8) via scaling by 2 and adding 4, i.e., 4 + 2*(U1 + U2).
However, you don't want a height of zero at the endpoints, you want half the peak's height. In other words, you want a triangle sitting on a rectangular base (i.e., uniform), with height h at the endpoints and height 2h in the middle. That makes life easy, because the triangle must have a peak of height h above the rectangular base, and a triangle with height h has half the area of a rectangle with the same base and height h. It follows that 2/3 of your probability is in the base, 1/3 is in the triangle.
Combining the elements above leads to the following pseudocode algorithm. If rnd() is a function call that returns continuous uniform(0,1) random numbers:
define makeValue()
if rnd() <= 2/3 # Caution, may want to use 2.0/3.0 for many languages
return 4 + (4 * rnd())
else
return 4 + (2 * (rnd() + rnd()))
I cranked out a million values using that and plotted a histogram:
For the case someone needs this in Game Maker (or a different language ) as an universal function:
if random(1) <= argument0
return argument1 + ((argument2-argument1) * random(1))
else
return argument1 + (((argument2-argument1)/2) * (random(1) + random(1)))
Called as follows (similar to the standard random_range function):
val = weight_random_range(FACTOR, FROM, TO)
"FACTOR" determines how much of the whole probability figure is the "base" for constant probability. E.g. 2/3 for the figure above.
0 will provide a perfect triangle and 1 a rectangle (no weightning).
Consider a discrete curve defined by the points (x1,y1), (x2,y2), (x3,y3), ... ,(xn,yn)
Define a constant SUM = y1+y2+y3+...+yn. Say we change the value of some k number of y points (increase or decrease) such that the total sum of these changed points is less than or equal to the constant SUM.
What would be the best possible manner to adjust the other y points given the following two conditions:
The total sum of the y points (y1'+y2'+...+yn') should remain constant ie, SUM.
The curve should retain as much of its original shape as possible.
A simple solution would be to define some delta as follows:
delta = (ym1' + ym2' + ym3' + ... + ymk') - (ym1 + ym2 + ym3 + ... + ymk')
and to distribute this delta over the rest of the points equally. Here ym1' is the value of the modified point after modification and ym1 is the value of the modified point before modification to give delta as the total difference in modification.
However this would not ensure a totally smoothed curve as area near changed points would appear ragged. Does a better solution/algorithm exist for the this problem?
I've used the following approach, though it is a bit OTT.
Consider adding d[i] to y[i], to get s[i], the smoothed value.
We seek to minimise
S = Sum{ 1<=i<N-1 | sqr( s[i+1]-2*s[i]+s[i-1] } + f*Sum{ 0<=i<N | sqr( d[i])}
The first term is a sum of the squares of (an approximate) second derivative of the curve, and the second term penalises moving away from the original. f is a (positive) constant. A little algebra recasts this as
S = sqr( ||A*d - b||)
where the matrix A has a nice structure, and indeed A'*A is penta-diagonal, which means that the normal equations (ie d = Inv(A'*A)*A'*b) can be solved efficiently. Note that d is computed directly, there is no need to initialise it.
Given the solution d to this problem we can compute the solution d^ to the same problem but with the constraint One'*d = 0 (where One is the vector of all ones) like this
d^ = d - (One'*d/Q) * e
e = Inv(A'*A)*One
Q = One'*e
What value to use for f? Well a simple approach is to try out this procedure on sample curves for various fs and pick a value that looks good. Another approach is to pick a estimate of smoothness, for example the rms of the second derivative, and then a value that should attain, and then search for an f that gives that value. As a general rule, the bigger f is the less smooth the smoothed curve will be.
Some motivation for all this. The aim is to find a 'smooth' curve 'close' to a given one. For this we need a measure of smoothness (the first term in S) and a measure of closeness (the second term. Why these measures? Well, each are easy to compute, and each are quadratic in the variables (the d[]); this will mean that the problem becomes an instance of linear least squares for which there are efficient algorithms available. Moreover each term in each sum depends on nearby values of the variables, which will in turn mean that the 'inverse covariance' (A'*A) will have a banded structure and so the least squares problem can be solved efficiently. Why introduce f? Well, if we didn't have f (or set it to 0) we could minimise S by setting d[i] = -y[i], getting a perfectly smooth curve s[] = 0, which has nothing to do with the y curve. On the other hand if f is gigantic, then to minimise s we should concentrate on the second term, and set d[i] = 0, and our 'smoothed' curve is just the original. So it's reasonable to suppose that as we vary f, the corresponding solutions will vary between being very smooth but far from y (small f) and being close to y but a bit rough (large f).
It's often said that the normal equations, whose use I advocate here, are a bad way to solve least squares problems, and this is generally true. However with 'nice' banded systems -- like the one here -- the loss of stability through using the normal equations is not so great, while the gain in speed is so great. I've used this approach to smooth curves with many thousands of points in a reasonable time.
To see what A is, consider the case where we had 4 points. Then our expression for S comes down to:
sqr( s[2] - 2*s[1] + s[0]) + sqr( s[3] - 2*s[2] + s[1]) + f*(d[0]*d[0] + .. + d[3]*d[3]).
If we substitute s[i] = y[i] + d[i] in this we get, for example,
s[2] - 2*s[1] + s[0] = d[2]-2*d[1]+d[0] + y[2]-2*y[1]+y[0]
and so we see that for this to be sqr( ||A*d-b||) we should take
A = ( 1 -2 1 0)
( 0 1 -2 1)
( f 0 0 0)
( 0 f 0 0)
( 0 0 f 0)
( 0 0 0 f)
and
b = ( -(y[2]-2*y[1]+y[0]))
( -(y[3]-2*y[2]+y[1]))
( 0 )
( 0 )
( 0 )
( 0 )
In an implementation, though, you probably wouldn't want to form A and b, as they are only going to be used to form the normal equation terms, A'*A and A'*b. It would be simpler to accumulate these directly.
This is a constrained optimization problem. The functional to be minimized is the integrated difference of the original curve and the modified curve. The constraints are the area under the curve and the new locations of the modified points. It is not easy to write such codes on your own. It is better to use some open source optimization codes, like this one: ool.
what about to keep the same dynamic range?
compute original min0,max0 y-values
smooth y-values
compute new min1,max1 y-values
linear interpolate all values to match original min max y
y=min1+(y-min1)*(max0-min0)/(max1-min1)
that is it
Not sure for the area but this should keep the shape much closer to original one. I got this Idea right now while reading your question and now I face similar problem so I try to code it and try right now anyway +1 for the getting me this Idea :)
You can adapt this and combine with the area
So before this compute the area and apply #1..#4 and after that compute new area. Then multiply all values by old_area/new_area ratio. If you have also negative values and not computing absolute area then you have to handle positive and negative areas separately and find multiplication ration to best fit original area for booth at once.
[edit1] some results for constant dynamic range
As you can see the shape is slightly shifting to the left. Each image is after applying few hundreds smooth operations. I am thinking of subdivision to local min max intervals to improve this ...
[edit2] have finished the filter for mine own purposes
void advanced_smooth(double *p,int n)
{
int i,j,i0,i1;
double a0,a1,b0,b1,dp,w0,w1;
double *p0,*p1,*w; int *q;
if (n<3) return;
p0=new double[n<<2]; if (p0==NULL) return;
p1=p0+n;
w =p1+n;
q =(int*)((double*)(w+n));
// compute original min,max
for (a0=p[0],i=0;i<n;i++) if (a0>p[i]) a0=p[i];
for (a1=p[0],i=0;i<n;i++) if (a1<p[i]) a1=p[i];
for (i=0;i<n;i++) p0[i]=p[i]; // store original values for range restoration
// compute local min max positions to p1[]
dp=0.01*(a1-a0); // min delta treshold
// compute first derivation
p1[0]=0.0; for (i=1;i<n;i++) p1[i]=p[i]-p[i-1];
for (i=1;i<n-1;i++) // eliminate glitches
if (p1[i]*p1[i-1]<0.0)
if (p1[i]*p1[i+1]<0.0)
if (fabs(p1[i])<=dp)
p1[i]=0.5*(p1[i-1]+p1[i+1]);
for (i0=1;i0;) // remove zeros from derivation
for (i0=0,i=0;i<n;i++)
if (fabs(p1[i])<dp)
{
if ((i> 0)&&(fabs(p1[i-1])>=dp)) { i0=1; p1[i]=p1[i-1]; }
else if ((i<n-1)&&(fabs(p1[i+1])>=dp)) { i0=1; p1[i]=p1[i+1]; }
}
// find local min,max to q[]
q[n-2]=0; q[n-1]=0; for (i=1;i<n-1;i++) if (p1[i]*p1[i-1]<0.0) q[i-1]=1; else q[i-1]=0;
for (i=0;i<n;i++) // set sign as +max,-min
if ((q[i])&&(p1[i]<-dp)) q[i]=-q[i]; // this shifts smooth curve to the left !!!
// compute weights
for (i0=0,i1=1;i1<n;i0=i1,i1++) // loop through all local min,max intervals
{
for (;(!q[i1])&&(i1<n-1);i1++); // <i0,i1>
b0=0.5*(p[i0]+p[i1]);
b1=fabs(p[i1]-p[i0]);
if (b1>=1e-6)
for (b1=0.35/b1,i=i0;i<=i1;i++) // compute weights bigger near local min max
w[i]=0.8+(fabs(p[i]-b0)*b1);
}
// smooth few times
for (j=0;j<5;j++)
{
for (i=0;i<n ;i++) p1[i]=p[i]; // store data to avoid shifting by using half filtered data
for (i=1;i<n-1;i++) // FIR smooth filter
{
w0=w[i];
w1=(1.0-w0)*0.5;
p[i]=(w1*p1[i-1])+(w0*p1[i])+(w1*p1[i+1]);
}
for (i=1;i<n-1;i++) // avoid local min,max shifting too much
{
if (q[i]>0) // local max
{
if (p[i]<p[i-1]) p[i]=p[i-1]; // can not be lower then neigbours
if (p[i]<p[i+1]) p[i]=p[i+1];
}
if (q[i]<0) // local min
{
if (p[i]>p[i-1]) p[i]=p[i-1]; // can not be higher then neigbours
if (p[i]>p[i+1]) p[i]=p[i+1];
}
}
}
for (i0=0,i1=1;i1<n;i0=i1,i1++) // loop through all local min,max intervals
{
for (;(!q[i1])&&(i1<n-1);i1++); // <i0,i1>
// restore original local min,max
a0=p0[i0]; b0=p[i0];
a1=p0[i1]; b1=p[i1];
if (a0>a1)
{
dp=a0; a0=a1; a1=dp;
dp=b0; b0=b1; b1=dp;
}
b1-=b0;
if (b1>=1e-6)
for (dp=(a1-a0)/b1,i=i0;i<=i1;i++)
p[i]=a0+((p[i]-b0)*dp);
}
delete[] p0;
}
so p[n] is the input/output data. There are few things that can be tweaked like:
weights computation (constants 0.8 and 0.35 means weights are <0.8,0.8+0.35/2>)
number of smooth passes (now 5 in the for loop)
the bigger the weight the less the filtering 1.0 means no change
The main Idea behind is:
find local extremes
compute weights for smoothing
so near local extremes are almost none change of the output
smooth
repair dynamic range per each interval between all local extremes
[Notes]
I did also try to restore the area but that is incompatible with mine task because it distorts the shape a lot. So if you really need the area then focus on that and not on the shape. The smoothing causes signal to shrink mostly so after area restoration the shape rise on magnitude.
Actual filter state has none markable side shifting of shape (which was the main goal for me). Some images for more bumpy signal (the original filter was extremly poor on this):
As you can see no visible signal shape shifting. The local extremes has tendency to create sharp spikes after very heavy smoothing but that was expected
Hope it helps ...
I am trying to understand the FFT algorithm and so far I think that I understand the main concept behind it. However I am confused as to the difference between 'framesize' and 'window'.
Based on my understanding, it seems that they are redundant with each other? For example, I present as input a block of samples with a framesize of 1024. So I have byte[1024] presented as input.
What then is the purpose of the windowing function? Since initially, I thought the purpose of the windowing function is to select the block of samples from the original data.
Thanks!
What then is the purpose of the windowing function?
It's to deal with so-called "spectral leakage": the FFT assumes an infinite series that repeats the given sample frame over and over again. If you have a sine wave that is an integral number of cycles within the sample frame, then all is good, and the FFT gives you a nice narrow peak at the proper frequency. But if you have a sine wave that is not an integral number of cycles, there's a discontinuity between the last and first sample, and the FFT gives you false harmonics.
Windowing functions lower the amplitudes at the beginning and the end of the sample frame, to reduce the harmonics caused by this discontinuity.
some diagrams from a National Instruments webpage on windowing:
integral # of cycles:
non-integer # of cycles:
for additional information:
http://www.tmworld.com/article/322450-Windowing_Functions_Improve_FFT_Results_Part_I.php
http://zone.ni.com/reference/en-XX/help/371361B-01/lvanlsconcepts/char_smoothing_windows/
http://www.physik.uni-wuerzburg.de/~praktiku/Anleitung/Fremde/ANO14.pdf
A rectangular window of length M has frequency response of sin(ω*M/2)/sin(ω/2), which is zero when ω = 2*π*k/M, for k ≠ 0. For a DFT of length N, where ω = 2*π*n/N, there are nulls at n = k * N/M. The ratio N/M isn't necessarily an integer. For example, if N = 40, and M = 32, then there are nulls at multiples of 1.25, but only the integer multiples will appear in the DFT, which is bins 5, 10, 15, and 20 in this case.
Here's a plot of the 1024-point DFT of a 32-point rectangular window:
M = 32
N = 1024
w = ones(M)
W = rfft(w, N)
K = N/M
nulls = abs(W[K::K])
plot(abs(W))
plot(r_[K:N/2+1:K], nulls, 'ro')
xticks(r_[:512:64])
grid(); axis('tight')
Note the nulls at every N/M = 32 bins. If N=M (i.e. the window length equals the DFT length), then there are nulls at all bins except at n = 0.
When you multiply a window by a signal, the corresponding operation in the frequency domain is the circular convolution of the window's spectrum with the signal's spectrum. For example, the DTFT of a sinusoid is a weighted delta function (i.e. an impulse with infinite height, infinitesimal extension, and finite area) located at the positive and negative frequency of the sinusoid. Convolving a spectrum with a delta function just shifts it to the location of the delta and scales it by the delta's weight. Therefore when you multiply a window by a sinusoid in the sample domain, the window's frequency response is scaled and shifted to the frequency of the sinusoid.
There are a couple of scenarios to examine regarding the length of a rectangular window. First let's look at the case where the window length is an integer multiple of the sinusoid's period, e.g. a 32-sample rectangular window of a cosine with a period of 32/8 = 4 samples:
x1 = cos(2*pi*8*r_[:32]/32) # ω0 = 8π/16, bin 8/32 * 1024 = 256
X1 = rfft(x1 * w, 1024)
plot(abs(X1))
xticks(r_[:513:64])
grid(); axis('tight')
As before, there are nulls at multiples of N/M = 32. But the window's spectrum has been shifted to bin 256 of the sinusoid and scaled by its magnitude, which is 0.5 split between the positive frequency and the negative frequency (I'm only plotting positive frequencies). If the DFT length had been 32, the nulls would line up at every bin, prompting the appearance that there's no leakage. But that misleading appearance is only a function of the DFT length. If you pad the windowed signal with zeros (as above), you'll get to see the sinc-like response at frequencies between the nulls.
Now let's look at a case where the window length is not an integer multiple of the sinusoid's period, e.g. a cosine with an angular frequency of 7.5π/16 (the period is 64 samples):
x2 = cos(2*pi*15*r_[:32]/64) # ω0 = 7.5π/16, bin 15/64 * 1024 = 240
X2 = rfft(x2 * w, 1024)
plot(abs(X2))
xticks(r_[-16:513:64])
grid(); axis('tight')
The center bin location is no longer at an integer multiple of 32, but shifted by a half down to bin 240. So let's see what the corresponding 32-point DFT would look like (inferring a 32-point rectangular window). I'll compute and plot the 32-point DFT of x2[n] and also superimpose a 32x decimated copy of the 1024-point DFT:
X2_32 = rfft(x2, 32)
X2_sample = X2[::32]
stem(r_[:17],abs(X2_32))
plot(abs(X2_sample), 'rs') # red squares
grid(); axis([0,16,0,11])
As you can see in the previous plot, the nulls are no longer aligned at multiples of 32, so the magnitude of the 32-point DFT is non-zero at each bin. In the 32 point DFT, the window's nulls are still spaced every N/M = 32/32 = 1 bin, but since ω0 = 7.5π/16, the center is at 'bin' 7.5, which puts the nulls at 0.5, 1.5, etc, so they're not present in the 32-point DFT.
The general message is that spectral leakage of a windowed signal is always present but can be masked in the DFT if the signal specrtum, window length, and DFT length come together in just the right way to line up the nulls. Beyond that you should just ignore these DFT artifacts and concentrate on the DTFT of your signal (i.e. pad with zeros to sample the DTFT at higher resolution so you can clearly examine the leakage).
Spectral leakage caused by convolving with a window's spectrum will always be there, which is why the art of crafting particularly shaped windows is so important. The spectrum of each window type has been tailored for a specific task, such as dynamic range or sensitivity.
Here's an example comparing the output of a rectangular window vs a Hamming window:
from pylab import *
import wave
fs = 44100
M = 4096
N = 16384
# load a sample of guitar playing an open string 6
# with a fundamental frequency of 82.4 Hz
g = fromstring(wave.open('dist_gtr_6.wav').readframes(-1),
dtype='int16')
L = len(g)/4
g_t = g[L:L+M]
g_t = g_t / float64(max(abs(g_t)))
# compute the response with rectangular vs Hamming window
g_rect = rfft(g_t, N)
g_hamm = rfft(g_t * hamming(M), N)
def make_plot():
fmax = int(82.4 * 4.5 / fs * N) # 4 harmonics
subplot(211); title('Rectangular Window')
plot(abs(g_rect[:fmax])); grid(); axis('tight')
subplot(212); title('Hamming Window')
plot(abs(g_hamm[:fmax])); grid(); axis('tight')
if __name__ == "__main__":
make_plot()
If you don't modify the sample values, and select the same length of data as the FFT length, this is equivalent to using a rectangular window, in which case the frame and the window are identical. However multiplying your input data by a rectangular window in the time domain is the same as convolving the input signal's spectrum with a Sinc function in the frequency domain, which will spread any spectral peaks for frequencies which are not exactly periodic in the FFT aperture across the entire spectrum.
Non-rectangular windows are often used so the the resulting FFT spectrum is convolved with something a bit more "focused" than a Sinc function.
You can also use a rectangular window that is a different size than the FFT length or aperture. In the case of a shorter data window, the FFT frame can be zero padded, which can result in an smoother looking interpolated FFT result spectrum. You can even use a rectangular window that is longer that the length of the FFT by wrapping data around the FFT aperture in a summed circular manner for some interesting effects with the frequency resolution.
ADDED due to a request:
Multiplying by a window in the time domain produces the same result as convolving with the transform of that window in the frequency domain.
In general, a narrower time domain window with produce a wider looking frequency domain transform. This is the reason that zero-padding produces a smoother frequency plot. The narrower time domain window produces a wider Sinc with fatter and smoother curves in relation to the frame width than would a window the full width of the FFT frame, thus making the interpolated frequency results look smoother than an non-zero padded FFT of the same frame length.
The converse is also true to some extent. A wider rectangular window will produce a narrower Sinc, with the nulls closer to the peak. Thus you might be able to use a carefully chosen wider window to produce a narrower looking Sinc to null a frequency closer to a bin of interest than 1 frequency bin away. How do you use a wider window? Wrap the data around and sum, which is identical to using FT basis vectors that are not truncated to 1 FFT frame in length. However, since when doing this the FFT result vector is shorter than the data, this is a lossy process which will introduce artifacts, and introduce some new novel aliasing. But it will give you a sharper frequency selection peak at each bin, and notch filters that can be placed less than 1 bin away, say halfway between bins, etc.
I want to know the frequency of data. I had a little bit idea that it can be done using FFT, but I am not sure how to do it. Once I passed the entire data to FFT, then it is giving me 2 peaks, but how can I get the frequency?
Thanks a lot in advance.
Here's what you're probably looking for:
When you talk about computing the frequency of a signal, you probably aren't so interested in the component sine waves. This is what the FFT gives you. For example, if you sum sin(2*pi*10x)+sin(2*pi*15x)+sin(2*pi*20x)+sin(2*pi*25x), you probably want to detect the "frequency" as 5 (take a look at the graph of this function). However, the FFT of this signal will detect the magnitude of 0 for the frequency 5.
What you are probably more interested in is the periodicity of the signal. That is, the interval at which the signal becomes most like itself. So most likely what you want is the autocorrelation. Look it up. This will essentially give you a measure of how self-similar the signal is to itself after being shifted over by a certain amount. So if you find a peak in the autocorrelation, that would indicate that the signal matches up well with itself when shifted over that amount. There's a lot of cool math behind it, look it up if you are interested, but if you just want it to work, just do this:
Window the signal, using a smooth window (a cosine will do. The window should be at least twice as large as the largest period you want to detect. 3 times as large will give better results). (see http://zone.ni.com/devzone/cda/tut/p/id/4844 if you are confused).
Take the FFT (however, make sure the FFT size is twice as big as the window, with the second half being padded with zeroes. If the FFT size is only the size of the window, you will effectively be taking the circular autocorrelation, which is not what you want. see https://en.wikipedia.org/wiki/Discrete_Fourier_transform#Circular_convolution_theorem_and_cross-correlation_theorem )
Replace all coefficients of the FFT with their square value (real^2+imag^2). This is effectively taking the autocorrelation.
Take the iFFT
Find the largest peak in the iFFT. This is the strongest periodicity of the waveform. You can actually be a little more clever in which peak you pick, but for most purposes this should be enough. To find the frequency, you just take f=1/T.
Suppose x[n] = cos(2*pi*f0*n/fs) where f0 is the frequency of your sinusoid in Hertz, n=0:N-1, and fs is the sampling rate of x in samples per second.
Let X = fft(x). Both x and X have length N. Suppose X has two peaks at n0 and N-n0.
Then the sinusoid frequency is f0 = fs*n0/N Hertz.
Example: fs = 8000 samples per second, N = 16000 samples. Therefore, x lasts two seconds long.
Suppose X = fft(x) has peaks at 2000 and 14000 (=16000-2000). Therefore, f0 = 8000*2000/16000 = 1000 Hz.
If you have a signal with one frequency (for instance:
y = sin(2 pi f t)
With:
y time signal
f the central frequency
t time
Then you'll get two peaks, one at a frequency corresponding to f, and one at a frequency corresponding to -f.
So, to get to a frequency, can discard the negative frequency part. It is located after the positive frequency part. Furthermore, the first element in the array is a dc-offset, so the frequency is 0. (Beware that this offset is usually much more than 0, so the other frequency components might get dwarved by it.)
In code: (I've written it in python, but it should be equally simple in c#):
import numpy as np
from pylab import *
x = np.random.rand(100) # create 100 random numbers of which we want the fourier transform
x = x - mean(x) # make sure the average is zero, so we don't get a huge DC offset.
dt = 0.1 #[s] 1/the sampling rate
fftx = np.fft.fft(x) # the frequency transformed part
# now discard anything that we do not need..
fftx = fftx[range(int(len(fftx)/2))]
# now create the frequency axis: it runs from 0 to the sampling rate /2
freq_fftx = np.linspace(0,2/dt,len(fftx))
# and plot a power spectrum
plot(freq_fftx,abs(fftx)**2)
show()
Now the frequency is located at the largest peak.
If you are looking at the magnitude results from an FFT of the type most common used, then a strong sinusoidal frequency component of real data will show up in two places, once in the bottom half, plus its complex conjugate mirror image in the top half. Those two peaks both represent the same spectral peak and same frequency (for strictly real data). If the FFT result bin numbers start at 0 (zero), then the frequency of the sinusoidal component represented by the bin in the bottom half of the FFT result is most likely.
Frequency_of_Peak = Data_Sample_Rate * Bin_number_of_Peak / Length_of_FFT ;
Make sure to work out your proper units within the above equation (to get units of cycles per second, per fortnight, per kiloparsec, etc.)
Note that unless the wavelength of the data is an exact integer submultiple of the FFT length, the actual peak will be between bins, thus distributing energy among multiple nearby FFT result bins. So you may have to interpolate to better estimate the frequency peak. Common interpolation methods to find a more precise frequency estimate are 3-point parabolic and Sinc convolution (which is nearly the same as using a zero-padded longer FFT).
Assuming you use a discrete Fourier transform to look at frequencies, then you have to be careful about how to interpret the normalized frequencies back into physical ones (i.e. Hz).
According to the FFTW tutorial on how to calculate the power spectrum of a signal:
#include <rfftw.h>
...
{
fftw_real in[N], out[N], power_spectrum[N/2+1];
rfftw_plan p;
int k;
...
p = rfftw_create_plan(N, FFTW_REAL_TO_COMPLEX, FFTW_ESTIMATE);
...
rfftw_one(p, in, out);
power_spectrum[0] = out[0]*out[0]; /* DC component */
for (k = 1; k < (N+1)/2; ++k) /* (k < N/2 rounded up) */
power_spectrum[k] = out[k]*out[k] + out[N-k]*out[N-k];
if (N % 2 == 0) /* N is even */
power_spectrum[N/2] = out[N/2]*out[N/2]; /* Nyquist freq. */
...
rfftw_destroy_plan(p);
}
Note it handles data lengths that are not even. Note particularly if the data length is given, FFTW will give you a "bin" corresponding to the Nyquist frequency (sample rate divided by 2). Otherwise, you don't get it (i.e. the last bin is just below Nyquist).
A MATLAB example is similar, but they are choosing the length of 1000 (an even number) for the example:
N = length(x);
xdft = fft(x);
xdft = xdft(1:N/2+1);
psdx = (1/(Fs*N)).*abs(xdft).^2;
psdx(2:end-1) = 2*psdx(2:end-1);
freq = 0:Fs/length(x):Fs/2;
In general, it can be implementation (of the DFT) dependent. You should create a test pure sine wave at a known frequency and then make sure the calculation gives the same number.
Frequency = speed/wavelength.
Wavelength is the distance between the two peaks.