Based on HEVC standard, decimal quantized coefficients are binarized through different methods, e.g. Truncated Unary, Truncated Rice (k-th order), Exp-Golomb, etc.
Considering the Truncated Rice method, I cannot understand the role of cmax parameter that completely changes the output. I have understood how the algorithm works, but I do not find any reference about that parameter. It seems that the Truncated Rice does not depend by cmax, however different outputs return for associated different values.
Comparing the two attached images, you can note the different binarized outputs based on cmax. Can you provide any reference that explains how the algorithm works based on cmax?
Could you add references to these screen shots, so we can read and understand the notation?
PS: I assume you have already read Transform Coefficient Coding in HEVC paper, but you still need more detailed information.
Related
When I use glm to perform logistic regression, first I read my binary outcome and predictor variables from a CSV file and convert them to factors (they are read in as ints). But if I do that before using gee or geeglm for clustering, geeglm gives me an error message, and although gee doesn't give me an error, the documentation says that you must use continuous predictors. It seems like I get sensical results if I just read in my binary variables and leave them as ints and do not convert them to factors. Is this an ok thing to do, and are there any concerns I need to be aware of? I specify binomial, and they are both for generalized linear models, so I'm not sure what's so wrong with passing factors.
Relatedly, in the scenario I'm describing, do I need to specify scale.fix=TRUE?
Thanks in advance!
I have a confusion about the way the LSTM networks work when forecasting with an horizon that is not finite but I'm rather searching for a prediction in whatever time in future. In physical terms I would call it the evolution of the system.
Suppose I have a time series $y(t)$ (output) I want to forecast, and some external inputs $u_1(t), u_2(t),\cdots u_N(t)$ on which the series $y(t)$ depends.
It's common to use the lagged value of the output $y(t)$ as input for the network, such that I schematically have something like (let's consider for simplicity just lag 1 for the output and no lag for the external input):
[y(t-1), u_1(t), u_2(t),\cdots u_N(t)] \to y(t)
In this way of thinking the network, when one wants to do recursive forecast it is forced to use the predicted value at the previous step as input for the next step. In this way we have an effect of propagation of error that makes the long term forecast badly behaving.
Now, my confusion is, I'm thinking as a RNN as a kind of an (simple version) implementation of a state space model where I have the inputs, my output and one or more state variable responsible for the memory of the system. These variables are hidden and not observed.
So now the question, if there is this kind of variable taking already into account previous states of the system why would I need to use the lagged output value as input of my network/model ?
Getting rid of this does my long term forecast would be better, since I'm not expecting anymore the propagation of the error of the forecasted output. (I guess there will be anyway an error in the internal state propagating)
Thanks !
Please see DeepAR - a LSTM forecaster more than one step into the future.
The main contributions of the paper are twofold: (1) we propose an RNN
architecture for probabilistic forecasting, incorporating a negative
Binomial likelihood for count data as well as special treatment for
the case when the magnitudes of the time series vary widely; (2) we
demonstrate empirically on several real-world data sets that this
model produces accurate probabilistic forecasts across a range of
input characteristics, thus showing that modern deep learning-based
approaches can effective address the probabilistic forecasting
problem, which is in contrast to common belief in the field and the
mixed results
In this paper, they forecast multiple steps into the future, to negate exactly what you state here which is the error propagation.
Skipping several steps allows to get more accurate predictions, further into the future.
One more thing done in this paper is predicting percentiles, and interpolating, rather than predicting the value directly. This adds stability, and an error assessment.
Disclaimer - I read an older version of this paper.
I want to fit a curve to data obtained from an FFT. While working on this, I remembered that an FFT gives binned data, and therefore I wondered if I should treat this differently with curve-fitting.
If the bins are narrow compared to the structure, I think it should not be necessary to treat the data differently, but for me that is not the case.
I expect the right way to fit binned data is by minimizing not the difference between values of the bin and fit, but between bin area and the area beneath the fitted curve, for each bin, such that the energy in each bin matches the energy in the range of the bin as signified by the curve.
So my question is: am I thinking correctly about this? If not, how should I go about it?
Also, when looking around for information about this subject, I encountered the "Maximum log likelihood" for example, but did not find enough information about it to understand if and how it applied to my situation.
PS: I have no clue if this is the right site for this question, please let me know if there is a better place.
For an unwindowed FFT, the correct interpolation between bins is by using a Sinc (sin(x)/x) or periodic Sinc (Dirichlet) interpolation kernel. For an FFT of samples of a band-limited signal, thus will reconstruct the continuous spectrum.
A very simple and effective way of interpolating the spectrum (from an FFT) is to use zero-padding. It works both with and without windowing prior to the FFT.
Take your input vector of length N and extend it to length M*N, where M is an integer
Set all values beyond the original N values to zeros
Perform an FFT of length (N*M)
Calculate the magnitude of the ouput bins
What you get is the interpolated spectrum.
Best regards,
Jens
This can be done by using maximum log likelihood estimation. This is a method that finds the set of parameters that is most likely to have yielded the measured data - the technique originates in statistics.
I have finally found an understandable source for how to apply this to binned data. Sadly I cannot enter formulas here, so I refer to that source for a full explanation: slide 4 of this slide show.
EDIT:
For noisier signals this method did not seem to work very well. A method that was a bit more robust is a least squares fit, where the difference between the area is minimized, as suggested in the question.
I have not found any literature to defend this method, but it is similar to what happens in the maximum log likelihood estimation, and yields very similar results for noiseless test cases.
The documentation for both of these methods are both very generic wherever I look. I would like to know what exactly I'm looking at with the returned arrays I'm getting from each method.
For getByteTimeDomainData, what time period is covered with each pass? I believe most oscopes cover a 32 millisecond span for each pass. Is that what is covered here as well? For the actual element values themselves, the range seems to be 0 - 255. Is this equivalent to -1 - +1 volts?
For getByteFrequencyData the frequencies covered is based on the sampling rate, so each index is an actual frequency, but what about the actual element values themselves? Is there a dB range that is equivalent to the values returned in the returned array?
getByteTimeDomainData (and the newer getFloatTimeDomainData) return an array of the size you requested - its frequencyBinCount, which is calculated as half of the requested fftSize. That array is, of course, at the current sampleRate exposed on the AudioContext, so if it's the default 2048 fftSize, frequencyBinCount will be 1024, and if your device is running at 44.1kHz, that will equate to around 23ms of data.
The byte values do range between 0-255, and yes, that maps to -1 to +1, so 128 is zero. (It's not volts, but full-range unitless values.)
If you use getFloatFrequencyData, the values returned are in dB; if you use the Byte version, the values are mapped based on minDecibels/maxDecibels (see the minDecibels/maxDecibels description).
Mozilla 's documentation describes the difference between getFloatTimeDomainData and getFloatFrequencyData, which I summarize below. Mozilla docs reference the Web Audio
experiment ; the voice-change-o-matic. The voice-change-o-matic illustrates the conceptual difference to me (it only works in my Firefox browser; it does not work in my Chrome browser).
TimeDomain/getFloatTimeDomainData
TimeDomain functions are over some span of time.
We often visualize TimeDomain data using oscilloscopes.
In other words:
we visualize TimeDomain data with a line chart,
where the x-axis (aka the "original domain") is time
and the y axis is a measure of a signal (aka the "amplitude").
Change the voice-change-o-matic "visualizer setting" to Sinewave to
see getFloatTimeDomainData(...)
Frequency/getFloatFrequencyData
Frequency functions (GetByteFrequencyData) are at a point in time; i.e. right now; "the current frequency data"
We sometimes see these in mp3 players/ "winamp bargraph style" music players (aka "equalizer" visualizations).
In other words:
we visualize Frequency data with a bar graph
where the x-axis (aka "domain") are frequencies or frequency bands
and the y-axis is the strength of each frequency band
Change the voice-change-o-matic "visualizer setting" to Frequency bars to see getFloatFrequencyData(...)
Fourier Transform (aka Fast Fourier Transform/FFT)
Another way to think about "time domain vs frequency" is shown the diagram below, from Fast Fourier Transform wikipedia
getFloatTimeDomainData gives you the chart on on the top (x-axis is Time)
getFloatFrequencyData gives you the chart on the bottom (x-axis is Frequency)
a Fast Fourier Transform (FFT) converts the Time Domain data into Frequency data, in other words, FFT converts the first chart to the second chart.
cwilso has it backwards.
the time data array is the longer one (fftSize), and the frequency data array is the shorter one (half that, frequencyBinCount).
fftSize of 2048 at the usual sample rate of 44.1kHz means each sample has 1/44100 duration, you have 2048 samples at hand, and thus are covering a duration of 2048/44100 seconds, which 46 milliseconds, not 23 milliseconds. The frequencyBinCount is indeed 1024, but that refers to the frequency domain (as the name suggests), not the time domain, and it the computation 1024/44100, in this context, is about as meaningful as adding your birth date to the fftSize.
A little math illustrating what's happening: Fourier transform is a 'vector space isomorphism', that is, a mapping going bijectively (i.e., reversible) between 2 vector spaces of the same dimension; the 'time domain' and the 'frequency domain.' The vector space dimension we have here (in both cases) is fftSize.
So where does the 'half' come from? The frequency domain coefficients 'count double'. Either because they 'actually are' complex numbers, or because you have the 'sin' and the 'cos' flavor. Or, because you have a 'magnitude' and a 'phase', which you'll understand if you know how complex numbers work. (Those are 3 ways to say the same in a different jargon, so to speak.)
I don't know why the API only gives us half of the relevant numbers when it comes to frequency - I can only guess. And my guess is that those are the 'magnitude' numbers, and the 'phase' numbers are thrown out. The reason that this is my guess is that in applications, magnitude is far more important than phase. Still, I'm quite surprised that the API throws out information, and I'd be glad if some expert who actually knows (and isn't guessing) can confirm that it's indeed the magnitude. Or - even better (I love to learn) - correct me.
I'd like to write a program that lets users draw points, lines, and circles as though with a straightedge and compass. Then I want to be able to answer the question, "are these three points collinear?" To answer correctly, I need to avoid rounding error when calculating the points.
Is this possible? How can I represent the points in memory?
(I looked into some unusual numeric libraries, but I didn't find anything that claimed to offer both exact arithmetic and exact comparisons that are guaranteed to terminate.)
Yes.
I highly recommend Introduction to constructions, which is a good basic guide.
Basically you need to be able to compute with constructible numbers - numbers that are either rational, or of the form a + b sqrt(c) where a,b,c were previously created (see page 6 on that PDF). This could be done with algebraic data type (e.g. data C = Rational Integer Integer | Root C C C in Haskell, where Root a b c = a + b sqrt(c)). However, I don't know how to perform tests with that representation.
Two possible approaches are:
Constructible numbers are a subset of algebraic numbers, so you can use algebraic numbers.
All algebraic numbers can be represented using polynomials of whose they are roots. The operations are computable, so if you represent a number a with polynomial p and b with polynomial q (p(a) = q(b) = 0), then it is possible to find a polynomial r such that r(a+b) = 0. This is done in some CASes like Mathematica, example. See also: Computional algebraic number theory - chapter 4
Use Tarski's test and represent numbers. It is slow (doubly exponential or so), but works :) Example: to represent sqrt(2), use the formula x^2 - 2 && x > 0. You can write equations for lines there, check if points are colinear etc. See A suite of logic programs, including Tarski's test
If you turn to computable numbers, then equality, colinearity etc. get undecidable.
I think the only way this would be possible is if you used a symbolic representation,
as opposed to trying to represent coordinate values directly -- so you would have
to avoid trying to coerce values like sqrt(2) into some numerical format. You will
be dealing with irrational numbers that are not finitely representable in binary,
decimal, or any other positional notation.
To expand on Jim Lewis's answer slightly, if you want to operate on points that are constructible from the integers with exact arithmetic, you will need to be able to operate on representations of the form:
a + b sqrt(c)
where a, b, and c are either rational numbers, or representations in the form given above. Wikipedia has a pretty decent article on the subject of what points are constructible.
Answering the question of exact equality (as necessary to establish colinearity) with such representations is a rather tricky problem.
If you try to compare co-ordinates for your points, then you have a problem. Leaving aside co-linearity for a moment, how about just working out whether two points are the same or not?
Supposing that one has given co-ordinates, and the other is a compass-straightedge construction starting from certain other co-ordinates, you want to determine with certainty whether they're the same point or not. Either way is a theorem of Euclidean geometry, it's not something you can just measure. You can prove they aren't the same by spotting some difference in their co-ordinates (for example by computing decimal places of each until you encounter a difference). But in general to prove they are the same cannot be done by approximate methods. Compute as many decimal places as you like of some expansions of 1/sqrt(2) and sqrt(2)/2, and you can prove they're very close together but you won't ever prove they're equal. That takes algebra (or geometry).
Similarly, to show that three points are co-linear you will need theorem-proving software. Represent the points A, B, C by their constructions, and attempt to prove the theorem "A, B and C are colinear". This is very hard - your program will prove some theorems but not others. Much easier is to ask the user for a proof that they are co-linear, and then verify (or refute) that proof, but that's probably not what you want.
In general, constructable points may have an arbitrarily complex symbolic form, so you must use a symbolic representation to work them exactly. As Stephen Canon noted above, you often need numbers of the form a+b*sqrt(c), where a and b are rational and c is an integer. All numbers of this form form a closed set under arithmetic operations. I have written some C++ classes (see rational_radical1.h) to work with these numbers if that is all you need.
It is also possible to construct numbers which are sums of any number of terms of rational multiples of radicals. When dealing with more than a single radicand, the numbers are no longer closed under multiplication and division, so you will need to store them as variable length rational coefficient arrays. The time complexity of operations will then be quadratic in the number of terms.
To go even further, you can construct the square root of any given number, so you could potentially have nested square roots. Here, the representations must be tree-like structures to deal with root hierarchy. While difficult to implement, there is nothing in principle preventing you from working with these representations. I'm not sure just what additional numbers can be constructed, but beyond a certain point, your symbolic representation will be expressive enough to handle very large classes of numbers.
Addendum
Found this Google Books link.
If the grid axes are integer valued then the answer is fairly straight forward, the points are either exactly colinear or they are not.
Typically however, one works with real numbers (well, floating points) and then draws the rounded values on the screen which does exist in integer space. In this case you have no choice but to pick a tolerance and use it to determine colinearity. Keep it small and the users will never know the difference.
You seem to be asking, in effect, "Can the normal mathematics (integer or floating point) used by computers be made to represent real numbers perfectly, with no rounding errors?" And, of course, the answer to that is "No." If you want theoretical correctness, then you will be stuck with the much harder problem of symbolic manipulation and coding up the equivalent of the inferences that are done in geometry. (In short, I'm agreeing with Steve Jessop, above.)
Some thoughts in the hope that they might help.
The sort of constructions you're talking about will require multiplication and division, which means that to preserve exactness you'll have to use rational numbers, which are generally easy to implement on top of a suitable sort of big integer (i.e., of unbounded magnitude). (Common Lisp has these built-in, and there have to be other languages.)
Now, you need to represent square roots of arbitrary numbers, and these have to be mixed in.
Therefore, a number is one of: a rational number, a rational number multiplied by a square root of a rational number (or, alternately, just the square root of a rational), or a sum of numbers. In order to prove anything, you're going to have to get these numbers into some sort of canonical form, which for all I can figure offhand may be annoying and computationally expensive.
This of course means that the users will be restricted to rational points and cannot use arbitrary rotations, but that's probably not important.
I would recommend no to try to make it perfectly exact.
The first reason for this is what you are asking here, the rounding error and all that stuff that comes with floating point calculations.
The second one is that you have to round your input as the mouse and screen work with integers. So, initially all user input would be integers, and your output would be integers.
Beside, from a usability point of view, its easier to click in the neighborhood of another point (in a line for example) and that the interface consider you are clicking in the point itself.