I have 3 arrays of X, Y and Z. Each have 8 elements. Now for each possible combination of (X,Y,Z) I have a V value.
I am looking to find a formula e.g. V=f(X,Y,Z). Any idea about how that can be done?
Thank you in advance,
Astry
You have a function sampled on a (possibly nonuniform) 3D grid, and want to evaluate the function at any arbitrary point within the volume. One way to approach this (some say the best) is as a multivariate spline evaluation. https://en.wikipedia.org/wiki/Multivariate_interpolation
First, you need to find which rectangular parallelepiped contains the (x,y,z) query point, then you need to interpolate the value from the nearest points. The easiest thing is to use trilinear interpolation from the nearest 8 points. If you want a smoother surface, you can use quadratic interpolation from 27 points or cubic interpolation from 64 points.
For repeated queries of a tricubic spline, your life would be a bit easier by preprocessing the spline to generate Hermite patches/volumes, where your sample points not only have the function value, but also its derivatives (∂/∂x, ∂/∂y, ∂/∂z). That way you don't need messy code for the boundaries at evaluation time.
Related
I am trying to implement nn.MultiheadAttention in my network. According to the docs,
embed_dim – total dimension of the model.
However, according to the source file,
embed_dim must be divisible by num_heads
and
self.q_proj_weight = Parameter(torch.Tensor(embed_dim, embed_dim))
If I understand properly, this means each head takes only a part of features of each query, as the matrix is quadratic. Is it a bug of realization or is my understanding wrong?
Each head uses a different part of the projected query vector. You can imagine it as if the query gets split into num_heads vectors that are independently used to compute the scaled dot-product attention. So, each head operates on a different linear combination of the features in queries (and keys and values, too). This linear projection is done using the self.q_proj_weight matrix and the projected queries are passed to F.multi_head_attention_forward function.
In F.multi_head_attention_forward, it is implemented by reshaping and transposing the query vector, so that the independent attentions for individual heads can be computed efficiently by matrix multiplication.
The attention head sizes are a design decision of PyTorch. In theory, you could have a different head size, so the projection matrix would have a shape of embedding_dim × num_heads * head_dims. Some implementations of transformers (such as C++-based Marian for machine translation, or Huggingface's Transformers) allow that.
All examples that I can find on the Internet just visualize the result array of the function computeSpectrum, but I am tasked with something else.
I generate a music note and I need by analyzing the result array to be able to say what note is playing. I figured out that I need to set the second parameter of the function call 'FFTMode' to true and then it returns sound frequencies. I thought that really it should return only one non-zero value which I could use to determine what note I generated using Math.sin function, but it is not the case.
Can somebody suggest a way how I can accomplish the task? Using the soundMixer.computeSpectrum is a requirement because I am going to analyze more complex sounds later.
FFT will transform your signal window into set of Nyquist sine waves so unless 440Hz is one of them you will obtain more than just one nonzero value! For a single sine wave you would obtain 2 frequencies due to aliasing. Here an example:
As you can see for exact Nyquist frequency the FFT response is single peak but for nearby frequencies there are more peaks.
Due to shape of the signal you can obtain continuous spectrum with peaks instead of discrete values.
Frequency of i-th sample is f(i)=i*samplerate/N where i={0,1,2,3,4,...(N/2)-1} is sample index (first one is DC offset so not frequency for 0) and N is the count of samples passed to FFT.
So in case you want to detect some harmonics (multiples of single fundamental frequency) then set the samplerate and N so samplerate/N is that fundamental frequency or divider of it. That way you would obtain just one peak for harmonics sinwaves. Easing up the computations.
Given a dataset of samples in a multi dimensional space (in my case a 4D space) where the samples are present on all the corners of the 4D cube and a substantial amount of samples within this cube but not in a neatly grid. Each sample has an output value next to it's 4D coordinate. The cube has coordinates [0,0,0,0]..[1,1,1,1].
Given a new coordinate (4D) how can I come up with the best interpolated value given these samples? Eg how do I choose the samples to start with, how to interpolate.
As a first guess I would guess that this can be done with a two step process:
find the smallest convex pentachoron (4D equivalent of the 3D tetrahedron / the 2D triangle) around the coordinate we need to interpolate.
interpolate within this tetrahedron.
Especially step 1 seems quite complex and slow.
Here's the first approach I'd try.
Step 1
Find the point's 4 nearest neighbors by Euclidean distance. It's important that these 4 points are linearly independent because next they're used to create a Barycentric coordinate system. Those 4 points become the vertices of your pentachoron (aka 4-simplex).
If nearest-neighbor checks are too slow, try structuring your data into a spatial lookup tree that works in 4D.
Step 2
Now we need to associate a value with the interpolation point X. Start by deriving X's representation in this new Barycentric coordinate system. This Barycentric coordinate consists of 4 numbers, which collectively describe the relative distance between the interpolation point and each of the 4-simplex's vertices.
Normalize the Barycentric coordinate so its components sum to 1.
Each of those 4 simplex vertices are data points and have an output value. Combine those 4 output values into a vector.
Finally, interpolate by calculating the dot product of the normalized coordinate with the vector of output values.
Source: This idea is really just a 4D extension of this gem in middle of the Barycentric coordinate system page on Wikipedia.
How should stereo (2 channel) audio data be represented for FFT? Do you
A. Take the average of the two channels and assign it to the real component of a number and leave the imaginary component 0.
B. Assign one channel to the real component and the other channel to the imag component.
Is there a reason to do one or the other? I searched the web but could not find any definite answers on this.
I'm doing some simple spectrum analysis and, not knowing any better, used option A). This gave me an unexpected result, whereas option B) went as expected. Here are some more details:
I have a WAV file of a piano "middle-C". By definition, middle-C is 260Hz, so I would expect the peak frequency to be at 260Hz and smaller peaks at harmonics. I confirmed this by viewing the spectrum via an audio editing software (Sound Forge). But when I took the FFT myself, with option A), the peak was at 520Hz. With option B), the peak was at 260Hz.
Am I missing something? The explanation that I came up with so far is that representing stereo data using a real and imag component implies that the two channels are independent, which, I suppose they're not, and hence the mess-up.
I don't think you're taking the average correctly. :-)
C. Process each channel separately, assigning the amplitude to the real component and leaving the imaginary component as 0.
Option B does not make sense. Option A, which amounts to convert the signal to mono, is OK (if you are interested in a global spectrum).
Your problem (double freq) is surely related to some misunderstanding in the use of your FFT routines.
Once you take the FFT you need to get the Magnitude of the complex frequency spectrum. To get the magnitude you take the absolute of the complex spectrum |X(w)|. If you want to look at the power spectrum you square the magnitude spectrum, |X(w)|^2.
In terms of your frequency shift I think it has to do with you setting the imaginary parts to zero.
If you imagine the complex Frequency spectrum as a series of complex vectors or position vectors in a cartesian space. If you took one discrete frequency bin X(w), there would be one real component representing its direction in the real axis (x -direction), and one imaginary component in the in the imaginary axis (y - direction). There are four important values about this discrete frequency, 1. real value, 2. imaginary value, 3. Magnitude and, 4. phase. If you just take the real value and set imaginary to 0, you are setting Magnitude = real and phase = 0deg or 90deg. You have hence forth modified the resulting spectrum, and applied a bias to every frequency bin. Take a look at the wiki on Magnitude of a vector, also called the Euclidean norm of a vector to brush up on your understanding. Leonbloy was correct, but I hope this was more informative.
Think of the FFT as a way to get information from a single signal. What you are asking is what is the best way to display data from two signals. My answer would be to treat each independently, and display an FFT for each.
If you want a really fast streaming FFT you can read about an algorithm I wrote here: www.depthcharged.us/?p=176
I want to invert a 4x4 matrix. My numbers are stored in fixed-point format (1.15.16 to be exact).
With floating-point arithmetic I usually just build the adjoint matrix and divide by the determinant (e.g. brute force the solution). That worked for me so far, but when dealing with fixed point numbers I get an unacceptable precision loss due to all of the multiplications used.
Note: In fixed point arithmetic I always throw away some of the least significant bits of immediate results.
So - What's the most numerical stable way to invert a matrix? I don't mind much about the performance, but simply going to floating-point would be to slow on my target architecture.
Meta-answer: Is it really a general 4x4 matrix? If your matrix has a special form, then there are direct formulas for inverting that would be fast and keep your operation count down.
For example, if it's a standard homogenous coordinate transform from graphics, like:
[ux vx wx tx]
[uy vy wy ty]
[uz vz wz tz]
[ 0 0 0 1]
(assuming a composition of rotation, scale, translation matrices)
then there's an easily-derivable direct formula, which is
[ux uy uz -dot(u,t)]
[vx vy vz -dot(v,t)]
[wx wy wz -dot(w,t)]
[ 0 0 0 1 ]
(ASCII matrices stolen from the linked page.)
You probably can't beat that for loss of precision in fixed point.
If your matrix comes from some domain where you know it has more structure, then there's likely to be an easy answer.
I think the answer to this depends on the exact form of the matrix. A standard decomposition method (LU, QR, Cholesky etc.) with pivoting (an essential) is fairly good on fixed point, especially for a small 4x4 matrix. See the book 'Numerical Recipes' by Press et al. for a description of these methods.
This paper gives some useful algorithms, but is behind a paywall unfortunately. They recommend a (pivoted) Cholesky decomposition with some additional features too complicated to list here.
I'd like to second the question Jason S raised: are you certain that you need to invert your matrix? This is almost never necessary. Not only that, it is often a bad idea. If you need to solve Ax = b, it is more numerically stable to solve the system directly than to multiply b by A inverse.
Even if you have to solve Ax = b over and over for many values of b, it's still not a good idea to invert A. You can factor A (say LU factorization or Cholesky factorization) and save the factors so you're not redoing that work every time, but you'd still solve the system each time using the factorization.
You might consider doubling to 1.31 before doing your normal algorithm. It'll double the number of multiplications, but you're doing a matrix invert and anything you do is going to be pretty tied to the multiplier in your processor.
For anyone interested in finding the equations for a 4x4 invert, you can use a symbolic math package to resolve them for you. The TI-89 will do it even, although it'll take several minutes.
If you give us an idea of what the matrix invert does for you, and how it fits in with the rest of your processing we might be able to suggest alternatives.
-Adam
Let me ask a different question: do you definitely need to invert the matrix (call it M), or do you need to use the matrix inverse to solve other equations? (e.g. Mx = b for known M, b) Often there are other ways to do this w/o explicitly needing to calculate the inverse. Or if the matrix M is a function of time & it changes slowly then you could calculate the full inverse once, & there are iterative ways to update it.
If the matrix represents an affine transformation (many times this is the case with 4x4 matrices so long as you don't introduce a scaling component) the inverse is simply the transpose of the upper 3x3 rotation part with the last column negated. Obviously if you require a generalized solution then looking into Gaussian elimination is probably the easiest.