In contrast to ffmpeg, it does not seem to be possible to adjust brightness or other "quick" settings of videos anymore using avconv, at least a grep in the manpage for brightness did not give a single result. Gamma correction seems to be hidden in some kind of LUT-filter.
Can anyone point me to some option (in ffmpeg, those where mp,eq2,later eq) that allows me to do so?
(On a sidenote, can anyone explain why this fundamental and useful functionality has been stripped or obfuscated to the user?)
Indeed as indicated in https://www.libav.org/avconv.html#lut_002c-lutrgb_002c-lutyuv you can use the following filters to change the gamma (use it with -vf):
lutyuv=y=gammaval(0.5)
or:
lutyuv="y=2*val"
If you are willing to play with RGB or YUV values you can probably get something better using formulas as in the examples of that website. For example, to increase saturation based on the formulas in https://stackoverflow.com/a/8810735/6040014:
lutyuv="y=2*val:u=(val-128)*2+128:v=(val-128)*2+128"
And an experiment with y=(val-128)*2+128 seems to get a contrast increase (but maybe "contrast" is a technical term that should follow better formulas than this).
Related
I have read several codes that do layer initialization using nn.init.kaiming_normal_() of PyTorch. Some codes use the fan in mode which is the default. Of the many examples, one can be found here and shown below.
init.kaiming_normal(m.weight.data, a=0, mode='fan_in')
However, sometimes I see people using the fan out mode as seen here and shown below.
if isinstance(m, nn.Conv2d):
nn.init.kaiming_normal_(m.weight, mode='fan_out', nonlinearity='relu')
Can someone give me some guidelines or tips to help me decide which mode to select? Further I am working on image super resolutions and denoising tasks using PyTorch and which mode will be more beneficial.
According to documentation:
Choosing 'fan_in' preserves the magnitude of the variance of the
weights in the forward pass. Choosing 'fan_out' preserves the
magnitudes in the backwards pass.
and according to Delving deep into rectifiers: Surpassing human-level performance on ImageNet classification - He, K. et al. (2015):
We note that it is sufficient to use either Eqn.(14) or
Eqn.(10)
where Eqn.(10) and Eqn.(14) are fan_in and fan_out appropriately. Furthermore:
This means that if the initialization properly scales the backward
signal, then this is also the case for the forward signal; and vice
versa. For all models in this paper, both forms can make them converge
so all in all it doesn't matter much but it's more about what you are after. I assume that if you suspect your backward pass might be more "chaotic" (greater variance) it is worth changing the mode to fan_out. This might happen when the loss oscillates a lot (e.g. very easy examples followed by very hard ones).
Correct choice of nonlinearity is more important, where nonlinearity is the activation you are using after the layer you are initializaing currently. Current defaults set it to leaky_relu with a=0, which is effectively the same as relu. If you are using leaky_relu you should change a to it's slope.
In Xmgrace, the var-epsilon (the inverted-3 form) can be written as \xe\f{}.
I need to write the epsilon (the lunate form) in xmgrace.
You can use the "\xÎ" combination to get something that is similar to (but not exactly) a lunate epsilon, but otherwise I am not aware of any other methods that can be used to directly input an \epsilon into xmgrace. Unfortunately, xmgrace is an old piece of software and does not have unicode support.
If you have no alternatives, I think that best course of action would be to generate an eps file and use inkscape or a similar software to directly edit the "right" epsilon in.
I didn't know before about artistic or artwork QR codes, while checking some of these codes, they are completely different from the regular standard QR code, but how is it possible to create this kind of QR code without loosing it's value (the scan result is the same) ?
These QR Codes are the most ones that amazed me:
http://www.hongkiat.com/blog/qr-code-artworks/
The only thing in common is the 3 corners, and they're different in style.
So my question is, what are the elements that we should preserve while creating such QR Codes ?
The most important things are:
Dark-on-light
Very nearly square modules
Modest light border
Substantially preserve the three-finder patterns
... and the first line of modules around them, which carries format info
... and the bottom-right alignment pattern, is helpful
The rest, the interior, can be substantially obscured and still be readable, certainly with high error correction. But messing with the elements above will tend to make it unreadable much more rapidly
The typical FFT for audio looks pretty similar to this, with most of the action happening on the far left side
http://www.flight404.com/blog/images/fft.jpg
He multiplied it by a partial sine wave to get it to the bottom, but the article isn't too specific on this part of it. It also seems like a "good enough" modification of the dataset, rather than one based on some property. I understand that human hearing is better suited to the higher frequencies, thus, most music will have amplified bass and attenuated treble so that both sound to us as being of relatively equal strength.
My question is what modification needs to be done to the FFT to compensate for this standard falloff?
for(i = 0; i < fft.length; i++){
fft[i] = fft[i] * Math.log(i + 1); // does, eh, ok but the high
// end is still not really "loud"
// enough
}
EDIT ::
http://en.wikipedia.org/wiki/Equal-loudness_contour
I came across this article, I think it might be the direction to head in, but there still might be some property of an FFT that needs to be counteracte.
First, are you sure you want to do this? It makes sense to compensate for some things, like the microphone response not being flat, but not human perception. People are used to hearing sounds with the spectral content that the sounds have in the real world, not along perceptual equal loudness curves. If you play a sound that you've modified in the way you suggest it would sound strange. Maybe some people like the music to have enhanced low frequencies, but this is a matter of taste, not psychophysics.
Or maybe you are compensating for some other reason, for example, taking into account the poorer sensitivity to lower frequencies might enhance a compression algorithm. Is this the idea?
If you do want to normalize by the equal loudness curves, one should note that most of the curves and equations are in terms of sound pressure level (SPL). SPL is the log of the square of the waveform amplitude, so when you work with the FFTs, it's probably easiest to work with their square (the power specta). (Or, of course, you could compensate in other ways by, say, multiplying by sqrt(log(i+1)) in your equation above -- assuming that the log was an approximation of the inverse equal-loudness curve.)
I think the equal loudness contour is exactly the right direction.
However, its shape depends on the absolute pressure level.
In other words the sensitivity curve of our hearing changes with sound pressure.
There is no "correct normalization" if you have no information about absolute levels.
If this is a problem depends on what you want to do with the data.
The loudness contour is standardized in ISO 226 but this document is not freely available for download. It should be in a decent university library though.
Here is another source for
loudness contours
So you are trying to raise the level of the high end frequencies? Sounds like a high pass filter with a minimum multiplier might work, so that you don't attenuate the low frequency signals too much. Pick up a good book on filter design, maybe monkey around with this applet
In the old days of first samplers, this is before MOTU Boost people :) it wasn't FFT but simple (Fairlight or Roland it first I think) Normalisation done on the original or resulting time-domain signal (if you are doing beat slicing, recycle-style); can't you do that? Or only go for the FFT after you compensate to counteract for it?
Seems like a two phase procedure otherwise, I'd personally leave FFT as is for the task..
What do you think of using a metric of function point to lines of code as a metric?
It makes me think of the old game show "Name That Tune". "I can name that tune in three notes!" I can write that functionality in 0.1 klocs! Is this useful?
It would certainly seem to promote library usage, but is that what you want?
I think it's a terrible idea. Just as bad as paying programmers by lines of code that they write.
In general, I prefer concise code over verbose code, but only as long as it still expresses the programmers' intention clearly. Maximizing function points per kloc is going to encourage everyone to write their code as briefly as they possibly can, which goes beyond concise and into cryptic. It will also encourage people to join adjacent lines of code into one line, even if said joining would not otherwise be desirable, just to reduce the number of lines of code. The maximum allowed line length would also become an issue.
KLOC is tolerable if you strictly enforce code standards, kind of like using page requirements for a report: no putting five statements on a single line or removing most of the whitespace from your code.
I guess one way you could decide how effective it is for your environment is to look at several different applications and modules, get a rough estimate of the quality of the code, and compare that to the size of the code. If you can demonstrate that code quality is consistent within your organization, then KLOC isn't a bad metric.
In some ways, you'll face the same battle with any similar metric. If you count feature or function points, or simply features or modules, you'll still want to weight them in some fashion. Ultimately, you'll need some sort of subjective supplement to the objective data you'll collect.
"What do you think of using a metric of function point to lines of code as a metric?"
Don't get the question. The above ratio is -- for a given language and team -- a simple statistical fact. And it tends toward a mean value with a small standard deviation.
There are lots of degrees of freedom: how you count function points, what language you're using, how (collectively) clever the team is. If you don't change those things, the value stays steady.
After a few projects together, you have a solid expectation that 1200 function points will be 12,000 lines of code in your preferred language/framework/team organization.
KSloc / FP is a bare statistical observation. Clearly, there's something else about this that's bothering you. Could you be more specific in your question?
The metric of Function Points to Lines of Code is actually used to generate the language level charts (actually, it is Function Points to Statements) to give an approximate sense of how powerful a programming language is. Here is an example: http://web.cecs.pdx.edu/~timm/dm/functionpoints.html
I wouldn't recommend using that ratio for anything else, except high level approximations like the language level chart.
Promoting library usage is a good thing, but the other thing to keep in mind is you will lose in the ratio when you are building the libraries, and will only pay it off with dividends of savings over time. Bean-counters won't understand that.
I personally would like to see a Function point to ABC metric ratio -- as I am curious about how the ABC metric (which indicates size and includes complexity as part of the info) would relate - perhaps linear, perhaps exponential, etc... www.softwarerenovation.com/ABCMetric.pdf
All metrics suck. My theory has always been that if you have to have them, then use the easiest thing you can to gather them and be done with it and onto important things.
That generally means something along the lines of
grep -c ";" *.h *.cpp | awk -F: '/:/ {x += $2} END {print x}'
If you are looking for a "metric" to track code efficency, don't. If you insist, again try something stupid but easy like source file size (see grep command above, w/o the awk pipe) or McCabe (with a counter program).