I am currently toying around with meshing voxel chunks, that is a N^3 list of voxel values. Since in my use-case most of the voxels are going to be of the same type a lot of the neighbors will share the same value. Thus, using RLE (run length encoding) makes sense to use, as it drastically cuts down the actual storage requirement at the cost of a O(log(n)) (n being the amount of runs in a chunk) random lookup. At the same time, I encode each voxel position as a z-order curve, specifically using morton encoding. This all works in my use-case and gives the expected space reductions required.
The issue is with meshing the individual chunks. Generating meshes takes up to 500ms (for N=48) which is simply too long for fluid gameplay. My current algorithm heavily borrows from the one described here and here.
Pseudocode algorithm:
- For each axis in [X, Y, Z]
- for each k in 0..N in axis
- Create a mask of size [N * N]
- for each u in 0..N of orthogonal axis:
- for each v in 0..N of second orthogonal axis:
- Check if (k, u, v) has a face between itself and (k+1, u, v)
- If yes, set mask[u*v*N] = true
- using the mask, make a greedy mesh and output that
The mesh generation itself is very fast (<<< 1ms) and wouldn't gain much from being optimized, but building the mask itself is very costly. As one can see in the tracing output below, each mesh_mask_building takes on average ~4ms which happens 3*N times per mesh!
My first thought to optimize this, was to use the inherent runs of the chunk and simply traverse those and build a mask of each, but this did not work out, as morton encoding is winding a lot throughout the primitive and as such would not be much better at constructing a mesh. It would also be highly suboptimal when one considers a chunk where only one layer is set to visible voxels. The current method generates a simple cuboid as it does not care about runs, but in my suggested method, each run would be seperate and generate more faces.
So my question is, how can I use a morton-run-length encoding to generate a mesh?
Related
In Karger's Min-Cut Algorithm for undirected (possibly weighted) multigraphs, the main operation is to contract a randomly chosen edge and merge it's incident vertices into one metavertex. This process is repeated until two vertices remain. These vertices correspond to a cut. The algo can be implemented with an adjacency list.
Questions:
how can I find the particular edge, that has been chosen to be contracted?
how does an edge get contracted (in an unweighted and/or weighted graph)?
Why does this procedure take quadratic time?
Edit: I have found some information that the runtime can be quadratic, due to the fact that we have O(n-2) contractions of vertices and each contraction can take O(n) time. It would be great if somebody could explain me, why a contraction takes linear time in an adjacency list. Note a contraction consists of: deleting one adjacent edge, merging the two vertices into a supernode, and making sure that the remaining adjacent edges are connected to the supernode.
Pseudocode:
procedure contract(G=(V,E)):
while |V|>2
choose edge uniformly at random
contract its endpoints
delete self loops
return cut
I have read the related topic Karger Min cut algorithm running time, but it did not help me. Also I do not have so much experience, so a "laymens" term explanation would be very much appreciated!
Minimally, I would like to know how to achieve what is stated in the title. Specifically, signal.lfilter seems like the only implementation of a difference equation filter in scipy, but it is 1D, as shown in the docs. I would like to know how to implement a 2D version as described by this difference equation. If that's as simple as "bro, use this function," please let me know, pardon my naiveté, and feel free to disregard the rest of the post.
I am new to DSP and acknowledging there might be a different approach to answering my question so I will explain the broader goal and give context for the question in the hopes someone knows how do want I want with Scipy, or perhaps a better way than what I explicitly asked for.
To get straight into it, broadly speaking I am using vectorized computation methods (Numpy/Scipy) to implement a Monte Carlo simulation to improve upon a naive for loop. I have successfully abstracted most of my operations to array computation / linear algebra, but a few specific ones (recursive computations) have eluded my intuition and I continually end up in the digital signal processing world when I go looking for how this type of thing has been done by others (that or machine learning but those "frameworks" are much opinionated). The reason most of my google searches end up on scipy.signal or scipy.ndimage library references is clear to me at this point, and subsequent to accepting the "signal" representation of my data, I have spent a considerable amount of time (about as much as reasonable for a field that is not my own) ramping up the learning curve to try and figure out what I need from these libraries.
My simulation entails updating a vector of data representing the state of a system each period for n periods, and then repeating that whole process a "Monte Carlo" amount of times. The updates in each of n periods are inherently recursive as the next depends on the state of the prior. It can be characterized as a difference equation as linked above. Additionally this vector is theoretically indexed on an grid of points with uneven stepsize. Here is an example vector y and its theoretical grid t:
y = np.r_[0.0024, 0.004, 0.0058, 0.0083, 0.0099, 0.0133, 0.0164]
t = np.r_[0.25, 0.5, 1, 2, 5, 10, 20]
I need to iteratively perform numerous operations to y for each of n "updates." Specifically, I am computing the curvature along the curve y(t) using finite difference approximations and using the result at each point to adjust the corresponding y(t) prior to the next update. In a loop this amounts to inplace variable reassignment with the desired update in each iteration.
y += some_function(y)
Not only does this seem inefficient, but vectorizing things seems intuitive given y is a vector to begin with. Furthermore I am interested in preserving each "updated" y(t) along the n updates, which would require a data structure of dimensions len(y) x n. At this point, why not perform the updates inplace in the array? This is wherein lies the question. Many of the update operations I have succesfully vectorized the "Numpy way" (such as adding random variates to each point), but some appear overly complex in the array world.
Specifically, as mentioned above the one involving computing curvature at each element using its neighbouring two elements, and then imediately using that result to update the next row of the array before performing its own curvature "update." I was able to implement a non-recursive version (each row fails to consider its "updated self" from the prior row) of the curvature operation using ndimage generic_filter. Given the uneven grid, I have unique coefficients (kernel weights) for each triplet in the kernel footprint (instead of always using [1,-2,1] for y'' if I had a uniform grid). This last part has already forced me to use a spatial filter from ndimage rather than a 1d convolution. I'll point out, something conceptually similar was discussed in this math.exchange post, and it seems to me only the third response saliently addressed the difference between mathematical notion of "convolution" which should be associative from general spatial filtering kernels that would require two sequential filtering operations or a cleverly merged kernel.
In any case this does not seem to actually address my concern as it is not about 2D recursion filtering but rather having a backwards looking kernel footprint. Additionally, I think I've concluded it is not applicable in that this only allows for "recursion" (backward looking kernel footprints in the spatial filtering world) in a manner directly proportional to the size of the recursion. Meaning if I wanted to filter each of n rows incorporating calculations on all prior rows, it would require a convolution kernel far too big (for my n anyways). If I'm understanding all this correctly, a recursive linear filter is algorithmically more efficient in that it returns (for use in computation) the result of itself applied over the previous n samples (up to a level where the stability of the algorithm is affected) using another companion vector (z). In my case, I would only need to look back one step at output signal y[n-1] to compute y[n] from curvature at x[n] as the rest works itself out like a cumsum. signal.lfilter works for this, but I can't used that to compute curvature, as that requires a kernel footprint that can "see" at least its left and right neighbors (pixels), which is how I ended up using generic_filter.
It seems to me I should be able to do both simultaneously with one filter namely spatial and recursive filtering; or somehow I've missed the maths of how this could be mathematically simplified/combined (convolution of multiples kernels?).
It seems like this should be a common problem, but perhaps it is rarely relevant to do both at once in signal processing and image filtering. Perhaps this is why you don't use signals libraries solely to implement a fast monte carlo simulation; though it seems less esoteric than using a tensor math library to implement a recursive neural network scan ... which I'm attempting to do right now.
EDIT: For those familiar with the theoretical side of DSP, I know that what I am describing, the process of designing a recursive filters with arbitrary impulse responses, is achieved by employing a mathematical technique called the z-transform which I understand is generally used for two things:
converting between the recursion coefficients and the frequency response
combining cascaded and parallel stages into a single filter
Both are exactly what I am trying to accomplish.
Also, reworded title away from FIR / IIR because those imply specific definitions of "recursion" and may be confusing / misnomer.
I have implemented an algorithm that uses two other algorithms for calculating the shortest path in a graph: Dijkstra and Bellman-Ford. Based on the time complexity of the these algorithms, I can calculate the running time of my implementation, which is easy giving the code.
Now, I want to experimentally verify my calculation. Specifically, I want to plot the running time as a function of the size of the input (I am following the method described here). The problem is that I have two parameters - number of edges and number of vertices.
I have tried to fix one parameter and change the other, but this approach results in two plots - one for varying number of edges and the other for varying number of vertices.
This leads me to my question - how can I determine the order of growth based on two plots? In general, how can one experimentally determine the running time complexity of an algorithm that has more than one parameter?
It's very difficult in general.
The usual way you would experimentally gauge the running time in the single variable case is, insert a counter that increments when your data structure does a fundamental (putatively O(1)) operation, then take data for many different input sizes, and plot it on a log-log plot. That is, log T vs. log N. If the running time is of the form n^k you should see a straight line of slope k, or something approaching this. If the running time is like T(n) = n^{k log n} or something, then you should see a parabola. And if T is exponential in n you should still see exponential growth.
You can only hope to get information about the highest order term when you do this -- the low order terms get filtered out, in the sense of having less and less impact as n gets larger.
In the two variable case, you could try to do a similar approach -- essentially, take 3 dimensional data, do a log-log-log plot, and try to fit a plane to that.
However this will only really work if there's really only one leading term that dominates in most regimes.
Suppose my actual function is T(n, m) = n^4 + n^3 * m^3 + m^4.
When m = O(1), then T(n) = O(n^4).
When n = O(1), then T(n) = O(m^4).
When n = m, then T(n) = O(n^6).
In each of these regimes, "slices" along the plane of possible n,m values, a different one of the terms is the dominant term.
So there's no way to determine the function just from taking some points with fixed m, and some points with fixed n. If you did that, you wouldn't get the right answer for n = m -- you wouldn't be able to discover "middle" leading terms like that.
I would recommend that the best way to predict asymptotic growth when you have lots of variables / complicated data structures, is with a pencil and piece of paper, and do traditional algorithmic analysis. Or possibly, a hybrid approach. Try to break the question of efficiency into different parts -- if you can split the question up into a sum or product of a few different functions, maybe some of them you can determine in the abstract, and some you can estimate experimentally.
Luckily two input parameters is still easy to visualize in a 3D scatter plot (3rd dimension is the measured running time), and you can check if it looks like a plane (in log-log-log scale) or if it is curved. Naturally random variations in measurements plays a role here as well.
In Matlab I typically calculate a least-squares solution to two-variable function like this (just concatenates different powers and combinations of x and y horizontally, .* is an element-wise product):
x = log(parameter_x);
y = log(parameter_y);
% Find a least-squares fit
p = [x.^2, x.*y, y.^2, x, y, ones(length(x),1)] \ log(time)
Then this can be used to estimate running times for larger problem instances, ideally those would be confirmed experimentally to know that the fitted model works.
This approach works also for higher dimensions but gets tedious to generate, maybe there is a more general way to achieve that and this is just a work-around for my lack of knowledge.
I was going to write my own explanation but it wouldn't be any better than this.
So I am aware that a convolution by FFT has a lower computational complexity than a convolution in real space. But what are the downsides of an FFT convolution?
Does the kernel size always have to match the image size, or are there functions that take care of this, for example in pythons numpy and scipy packages? And what about anti-aliasing effects?
FFT convolutions are based on the convolution theorem, which states that given two functions f and g, if Fd() and Fi() denote the direct and inverse Fourier transform, and * and . convolution and multiplication, then:
f*g = Fi(Fd(d).Fd(g))
To apply this to a signal f and a kernel g, there are some things you need to take care of:
f and g have to be of the same size for the multiplication step to be possible, so you need to zero-pad the kernel (or input, if the kernel is longer than it).
When doing a DFT, which is what FFT does, the resulting frequency domain representation of the function is periodic. This means that, by default, your kernel wraps around the edge when doing the convolution. If you want this, then all is great. But if not, you have to add an extra zero-padding the size of the kernel to avoid it.
Most (all?) FFT packages only work well (performance-wise) with sizes that do not have any large prime factors. Rounding the signal and kernel size up to the next power of two is a common practice that may result in a (very) significant speed-up.
If your signal and kernel sizes are f_l and g_l, doing a straightforward convolution in time domain requires g_l * (f_l - g_l + 1) multiplications and (g_l - 1) * (f_l - g_l + 1) additions.
For the FFT approach, you have to do 3 FFTs of size at least f_l + g_l, as well as f_l + g_l multiplications.
For large sizes of both f and g, the FFT is clearly superior with its n*log(n) complexity. For small kernels, the direct approach may be faster.
scipy.signal has both convolve and fftconvolve methods for you to play around. And fftconvolve handles all the padding described above transparently for you.
While fast convolution has better "big O" complexity than direct form convolution; there are a few drawbacks or caveats. I did some thinking about this topic for an article I wrote a while back.
Better "big O" complexity is not always better. Direct form convolution can be faster than using FFTs for filters smaller than a certain size. The exact size depends on the platform and implementations used. The crossover point is usually in the 10-40 coefficient range.
Latency. Fast convolution is inherently a blockwise algorithm. Queueing up hundreds or thousands of samples at a time before transforming them may be unacceptable for some real-time applications.
Implementation complexity. Direct form is simpler in terms of the memory, code space and in the theoretical background of the writer/maintainer.
On a fixed point DSP platform (not a general purpose CPU): the limited word size considerations of fixed-point FFT make large fixed point FFTs nearly useless. At the other end of the size spectrum, these chips have specialized MAC intstructions that are well designed for performing direct form FIR computation, increasing the range over which te O(N^2) direct form is faster than O(NlogN). These factors tend to create a limited "sweet spot" where fixed point FFTs are useful for Fast Convolution.
so I am making a simple simulation of different planets with individual velocity flying around space and orbiting each other.
I plan to simulate their pull on each other by considering each planet as projecting their own "gravity vector field." Each time step I'm going to add the vectors outputted from each planets individual vector field equation (V = -xj + (-yj) or some notation like it) except the one being effected in the calculation, and use the effected planets position as input to the equations.
However this would inaccurate, and does not consider the gravitational pull as continuous and constant. Bow do I calculate the movement of my planets if each is continuously effecting the others?
Thanks!
In addition to what Blender writes about using Newton's equations, you need to consider how you will be integrating over your "acceleration field" (as you call it in the comment to his answer).
The easiest way is to use Euler's Method. The problem with that is it rapidly diverges, but it has the advantage of being easy to code and to be reasonably fast.
If you are looking for better accuracy, and are willing to sacrifice some performance, one of the Runge-Kutta methods (probably RK4) would ordinarily be a good choice. I'll caution you that if your "acceleration field" is dynamic (i.e. it changes over time ... perhaps as a result of planets moving in their orbits) RK4 will be a challenge.
Update (Based on Comment / Question Below):
If you want to calculate the force vector Fi(tn) at some time step tn applied to a specific object i, then you need to compute the force contributed by all of the other objects within your simulation using the equation Blender references. That is for each object, i, you figure out how all of the other objects pull (apply force) and those vectors when summed will be the aggregate force vector applied to i. Algorithmically this looks something like:
for each object i
Fi(tn) = 0
for each object j ≠i
Fi(tn) = Fi(tn) + G * mi * mj / |pi(tn)-pj(tn)|2
Where pi(tn) and pj(tn) are the positions of objects i and j at time tn respectively and the | | is the standard Euclidean (l2) normal ... i.e. the Euclidean distance between the two objects. Also, G is the gravitational constant.
Euler's Method breaks the simulation into discrete time slices. It looks at the current state and in the case of your example, considers all of the forces applied in aggregate to all of the objects within your simulation and then applies those forces as a constant over the period of the time slice. When using
ai(tn) = Fi(tn)/mi
(ai(tn) = acceleration vector at time tn applied to object i, Fi(tn) is the force vector applied to object i at time tn, and mi is the mass of object i), the force vector (and therefore the acceleration vector) is held constant for the duration of the time slice. In your case, if you really have another method of computing the acceleration, you won't need to compute the force, and can instead directly compute the acceleration. In either event, with the acceleration being held as constant, the position at time tn+1, p(tn+1) and velocity at time tn+1, v(tn+1), of the object will be given by:
pi(tn+1) = 0.5*ai(tn)*(tn+1-tn)2 + vi(tn)*(tn+1-tn)+pi(tn)
vi(tn+1) = ai(tn+1)*(tn+1-tn) + vi(tn)
The RK4 method fits the driver of your system to a 2nd degree polynomial which better approximates its behavior. The details are at the wikipedia site I referenced above, and there are a number of other resources you should be able to locate on the web. The basic idea is that instead of picking a single force value for a particular timeslice, you compute four force vectors at specific times and then fit the force vector to the 2nd degree polynomial. That's fine if your field of force vectors doesn't change between time slices. If you're using gravity to derive the vector field, and the objects which are the gravitational sources move, then you need to compute their positions at each of the four sub-intervals in order compute the force vectors. It can be done, but your performance is going to be quite a bit poorer than using Euler's method. On the plus side, you get more accurate motion of the objects relative to each other. So, it's a challenge in the sense that it's computationally expensive, and it's a bit of a pain to figure out where all the objects are supposed to be for your four samples during the time slice of your iteration.
There is no such thing as "continuous" when dealing with computers, so you'll have to approximate continuity with very small intervals of time.
That being said, why are you using a vector field? What's wrong with Newton?
And the sum of the forces on an object is that above equation. Equate the two and solve for a
So you'll just have to loop over all the objects one by one and find the acceleration on it.