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There is another question with this same title, but the question is asked differently than what's troubling me, and the answer is not sufficient.
The most prominent analogies I hear to explain bandwidth are the highway example, and the pipe example. In the highway example, bandwidth is the amount of cars that can drive on the highway in a given amount of time, and in the pipe its an amount of water that can flow through.
My question is - by measuring by cars per second, or liters per second, does that mean that a longer highway, pipe or copper wire has a higher bandwidth than a shorter one? That seems strange to me.
Wouldn't it make more sense to give the highway bandwidth as the amount of lanes it has - irrespective of a unit of time? It just makes more sense to me and is simpler to say that the pipe is "1 foot in diameter" rather than "it carries 100 litres per second".
Why do we measure bandwidth in bits per second and not just in bits?
"My question is - by measuring by cars per second, or liters per second, does that mean that a longer highway, pipe or copper wire has a higher bandwidth than a shorter one?"
No!
Bandwidth is not about how many cars can fit on the road. It's about how many cars can pass a point on the road during a certain time. How many cars per second can pass under a bridge, for example.
No, it wouldn't. You quote a highway in terms of lanes, because it's more understandable, and a reasonable approximation to assume 4 lanes = 4x as much traffic. But even then, you might have a traffic jam, and then 4 lanes is 'transmitting' fewer cars per minute than it would otherwise.
With a hose pipe, the width of the pipe is the speed of transmission if you assume the same water pressure.
These assumptions don't apply to communications - when I'm transmitting 'a bit' nothing physical is moving *. A 'bit' is the smallest piece that 'information' can be broken down into, and in order to transmit it, something needs to change.
If I turn on my torch and shine it at you, I've sent one 'message' (my torch is on). To send you anything more detailed, I would need to turn it off and on again - morse code is an example of doing this. The pattern of switching it off and on gives you some letters. How fast I can switch it off and on again, is how fast I can send a message.
So it is with bandwidth. I need to change things to communicate. If I can change things faster, I can communicate faster.
"bits" would be a measure of the number of torches I own. Bits per second is how fast I can flick them on and off to send a message.
* Electrons and photons do move, as does air to carry sound. But the signal isn't the thing moving - I don't have to move an atom of air from my mouth to your ear to 'talk' to you, the wave propagates through the medium.
Related
I run an infectious disease spread model similar to "VIRUS" model in the model library changing the "infectiousness".
I did 20 runs each for infectiousness values 98% , 95% , 93% and the Maximum infected count was 74.05 , 73 ,78.9 respectively. (peak was at tick 38 for all 3 infectiousness values)
[I took the average of the infected count for each tick and took the maximum of these averages as the "maximum infected".]
I was expecting the maximum infected count to decrease when the infectiousness is reduced, but it didn't. As per what I understood this happens, because I considered the average values of each simulation run. (It is like I am considering a new simulation run with average infected count for each tick ).
I want to say that, I am considering all 20 simulation runs. Is there a way to do that other than the way I used the average?
In the Models Library Virus model with default parameter settings at other values, and those high infectiousness values, what I see when I run the model is a periodic variation in the numbers three classes of person. Look at the plot in the lower left corner, and you'll see this. What is happening, I believe, is this:
When there are many healthy, non-immune people, that means that there are many people who can get infected, so the number of infected people goes up, and the number of healthy people goes down.
Soon after that, the number of sick, infectious people goes down, because they either die or become immune.
Since there are now more immune people, and fewer infectious people, the number of non-immune healthy grows; they are reproducing. (See "How it works" in the Info tab.) But now we have returned to the situation in step 1, ... so the cycle continues.
If your model is sufficiently similar to the Models Library Virus model, I'd bet that this is part of what's happening. If you don't have a plot window like the Virus model, I recommend adding it.
Also, you didn't say how many ticks you are running the model for. If you run it for a short number of ticks, you won't notice the periodic behavior, but that doesn't mean it hasn't begun.
What this all means that increasing infectiousness wouldn't necessarily increase the maximum number infected: a faster rate of infection means that the number of individuals who can infected drops faster. I'm not sure that the maximum number infected over the whole run is an interesting number, with this model and a high infectiousness value. It depends what you are trying to understand.
One of the great things about NetLogo and some other ABM systems is that you can watch the system evolve over time, using various tools such as plots, monitors, etc. as well as just looking at the agents move around or change states over time. This can help you understand what is going on in a way that a single number like an average won't. Then you can use this insight to figure out a more informative way of measuring what is happening.
Another model where you can see a similar kind of periodic pattern is Wolf-Sheep Predation. I recommend looking at that. It may be easier to understand the pattern. (If you are interested in mathematical models of this kind of phenomenon, look up Lotka-Volterra models.)
(Real virus transmission can be more complicated, because a person (or other animal) is a kind of big "island" where viruses can reproduce quickly. If they reproduce too quickly, this can kill the host, and prevent further transmission of the virus. Sometimes a virus that reproduces more slowly can harm more people, because there is time for them to infect others. This blog post by Elliott Sober gives a relatively simple mathematical introduction to some of the issues involved, but his simple mathematical models don't take into account all of the complications involved in real virus transmission.)
EDIT: You added a comment Lawan, saying that you are interested in modeling COVID-19 transmission. This paper, Variation and multilevel selection of SARS‐CoV‐2 by Blackstone, Blackstone, and Berg, suggests that some of the dynamics that I mentioned in the preceding remarks might be characteristic of COVID-19 transmission. That paper is about six months old now, and it offered some speculations based on limited information. There's probably more known now, but this might suggest avenues for further investigation.
If you're interested, you might also consider asking general questions about virus transmission on the Biology Stackexchange site.
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I am quite stuck on a board, or something that could fit my needs.
I made a dual Projector 3D System at home, like this : http://www.cinema3dglass.com/Dual_projector_3D_polarization_system.php
And the HDMI can have different (8) formats(Left Right, Above Below, ect..) all here: https://www.tridef.com/user-guide/3d-file-formats
So the needed images on the incoming HDMI port can ble placed like above, and I need to split them on two separate HDMI outputs according to the format, so I can plug them into the projectors.
Basically I need a device that is in the first image in the first link above title: "HDMI Distribution Amplifier & EDID Emulator"
I know an Arduino can't handle this amount of processing, because I overloaded it with simlier tasks.
Can anyone Help me where to start? I foud Panda development board but that's too expensive.
Or if there is a not owerly expensive device existing for this task, I could buy that.
I manadged to use the system from Tridef 3D but that's hard to get working.
I'd like my device to get the input from a Chromecast 2.0, but if it's not possible a normal player 'll do it.
I found some devices called HDMI Demultiplexer they simply cut one half of the input, but that's quite expensieve, for 260$ and two would be needed.
Help me please.
Thanks in advance.
From the HDMI specification at page 56 the transfer/interconnection looks like this:
I would start with interleaved left/right format where even pixels are left and odd pixels are right because there is high chance that it does not need any FIFO. If you want standard left/right then you need single line FIFO for each channel and for up/down full image FIFO. In case variable clock is supported by your HW then this simplified example should work:
You need to add H/V sync decoder from Channel0 to reset the binary counter. Binary counter counts which data address is being processed. The single data-line to the AND gates should be D1 half of the input clock but not entirely you need to toggle between D0 and D1 depends in the timing of data processed (for pixels it would be D1 and for other data it would be D0) that is the variable clock I mentioned before. The comparator just compares the address against predefined constants (like half of line for non interleaved left/right format or detect even odd for interleaved format but both must take +/- other data offsets) beware the transfer is on bits not Bytes so the address will be multiplied by number of bits per data chunk ... The gates just toggle clock between left and right part. LATCHES make sure output signal will be not mixed and also boost the signal.
I would start with oscilloscope measurements of the channels so you can see how the data is transfered and then experiment-ate. If you use FPGA then you do not need to make any changes to the board while ecxperimentating with configurations as the circuit will be solely inside FPGA.
If variable clock is not supported then you need to use FIFO and or RAM to store the full line/image and then send the appropriate parts to their connectors. For that you most likely need full decoding capability so use the SIL9134 + SIL9135. Halving resolution will introduce timing problems because you will need more time to send half speed half frame then the full speed full frame (the auxiliary and sync data is copied not halved). If the sending has big enough gaps you could fit the missing time there but again not all HW can support it losing sync/flickering/etc. In such case you could change the resolution to a bit smaller (after halving) to fit in the send time ... or enlarge thi full resolution input (in x axis).
Good luck with your quest.
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I was trying to find something original and fun to do with artificial neural networks (ANNs) as a personal/learning project and I though it would be cool if I could predict the results of sports games (especially NHL games).
I'm pretty sure it would be easy to evolve an ANN that can predict which team is most likely to win (usually the team with the better record). However, what I would like to do is create an ANN that would tell how likely the outcome is, similar to bookmaker odds.
Is this something an ANN can do? In the affirmative, what kind of success can I expect? I know I can't beat the bookmaker (at least not with a software solution). I want do this as a recreational project/challenge to myself. I don't expect to bet money on sports games with this project.
Way back in the days of the IBM XT I played with a shareware ANN program to try and improve my chances on the British football (soccer) pools. This is a form of betting where you try and predict which football matches will result in draws. I assigned each team a number then looked back thorough past results and from them generated a single digit for the result. From memory it was 0 from a home win , 1 for an away win and 2 for a draw. Each result went on a single line in a training file. I would then run the training file through the program and generate the ANN settings. I would then look up the following Saturdays matches and feed them into the ANN then look for matches predicted as draws.
As the weeks went on my predictions of draws did definetly become more and more accurate. However ...
1) The XT was so slow that by Christmas it was taking 24 hours to generate the ANN settings from the training data. I really had better things to do with my precious (and expensive) PC.
2) Although it was better at predicting draws it wasn't predicting enough to actually win any money. Looking back I suppose the program had just worked out that Manchester United would always beat Sheffield United. This was more football knowledge than I had but not enough to win any money.
3) Entering the results into the training data and then generating the forthcoming matches data was taking me ages and to be honest sport bores me rigid.
So I gave up and didn't become a millionaire.
These days however PC's are much faster and much of the training data could be scraped from the web. But I still doubt it is a route to a fortune but its certainly an interesting project.
Ian
A reply above stated:
I know that if the bookmakers odds could be beaten by an ANN,
bookmakers would already be using one to fix their odds.
Bookmakers don't set the line based on their analysis of the teams - they set it based on their analysis of the betting public's opinion of the teams. An ideal line for the bookie is where he has exactly the same amount bet on each side of the line - then he is guaranteed a profit = the 'juice' on the losers' bets. They move the line as game approaches to try to keep that 50/50 split. Bookie may think Home team -5 is accurate line based on game analysis, but if he expects that will draw 2x $$ on the Home team he will not set the line at -5 - he will set at -7 or -8 - to where he expects to draw equal $$ for both -5 and +5 bets.
ANNs are really good at pattern matching and prediction, so yes, odds are you could build an ANN that does what you want.
You'll need more than just team win/loss ratio to make it really effective however. Feed it stats for the players, too. For real effectiveness, try to include game-flow information... like which players are on the line for each play (for football, for example).
Ultimately, the biggest problem you'll run into (aside from the whole "writing the ANN" issue) is getting the data you need to feed it.
I've done some stock market predictions with an AI and my conclusion is that it is not very hard to make an AI that gets good results with the historical data.
Making winning transactions in the future is a different ballgame.
I have just worked on this very problem (predicting English Premier League games) for the past 10 days, and ended up with very similar results using 3 different methods: SVM, Logistic Regression, and NN.
LR and NN will give probabilities. SVM outputs 0/1 (but it can be tweaked for probas too (I haven't tried yet).
I needed a "massive" (by my standards at least) feature set though (almost 300) and a good chunk of data (13 years worth).
Re. data, I got it from the web, simply.
Conclusion: I can just about match the bookies in terms of accuracy (predicting victories in my case). If I add the pre-match odds to the feature set, I get the exact same accuracy as the bookies (as expected), but no better (surely meaning my feature set is summarized in the bookies odds, and they have a little extra knowledge on top).
I'm sure there is a way to get better accuracy, either by improving the algos, or more likely by having extremely granular data (as in which players play which games, for how many minutes, and a lot of player-level historical stats, so as to build bottom-up models of team performance).
But bottom line is I can testify NNs work quite well for that purpose. SVM is slightly better though, in my limited experience.
I think it's indeed all about data, but there's no end to what you could feed it with in order to be more accurate : winning/loosing streaks, players biorhythms, player's girlfriends mood before the game, minor/major injuries they suffered in the recent past, extra-sportive events that are bothering the players, etc, etc, etc.
But I don't think you can accurately predict which team is more likely to win, it would be just a more-or-less educated guess.
In my opinion and experience, because of the excessively large number of factors in play, designing and training the ANN will be unreasonably complex and time-consuming. ANNs are good at pattern matching, and game prediction takes much deductive reasoning rather than mere pattern matching.
But if you want to enjoy learning neural networks, it will be a good adventure. If you are successful, you might want to host your code somewhere for others to see and learn!
For game prediction, it would be much easier and faster with decision trees or a rules engine and so on. This will be no easy task either, but it will be another interesting activity.
My belief is that the unpredictability of an event is due to lack of information and understanding...If you have all the knowledge, then yes it could be done. Or, the more knowledge you have, the better it can be done.
So in theory, the answer is yes.
However, in practice, you can get a PhD and have a whole career working on this question and you still may not succeed.
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Is LOC correct parameter for project estimation?
there are so many scenarios where complexity takes much more time for a single line of code,
other than LOC what could be the suggested parameter for project estimation?
As peoples are talking about functional point of program does it mean for use case related information?
i am trying to find out any solid base for full software developement estimation which can consist analysis, design, testcase preparation, and coding, please suggest?
Steve McConnell in Rapid Development (Microsoft Press, 1996):
Because different programming
languages produce such different bangs
for a given number of lines of code,
much of the software industry is
moving toward a measure called
"function points" to estimate program
sizes. A function point is a synthetic
measure of program size that is based
on a weighted sum of the number of
inputs, outputs, inquiries, and files.
Function points are useful because
they allow you to think about program
size in a languageindependent way.
Google "Function Point" for more information.
Seeing as developers are likely to* spend most of their time trying to test changes, lines-of-code is never a good indicator of size of a problem.
Let's suppose you have an existing large application - changing a single line of code may seem trivial, but the test planning and execution could take weeks.
Likewise, adding a relatively large amount of code in a single limited-scope module which is easily testable might be only a few days.
* they should do, at least. If they're spending more time writing code than testing it, it is probably full of bugs. And I mean BEFORE it reaches your dedicated QA team.
Only if you use it in the inverse.
-- Edit
But no. It isn't. It's a mostly useless measure, and generally harmful. As you note, less code is almost always better.
Other things to check? Well, what are you trying to measure? What result do you want to see from a change in the things that you would be checking? What sort of decisions will you be making on the basis of these changes?
LOC is one proxy measure for measuring the problem size.
LOC estimate can be used, and LOC count is relatively cheap to measure from historical projects. But LOC can be problematic if used for anything else than a proxy for problem size, as already pointed out by other answers.
Problem size is rather constant given the requirements. From a size estimate you can go to effort, schedule and cost estimates. It depends on your planning drivers such as cost or schedule. From the historical data you can find correlation how problem size translates to effort and how other planning drivers further influence the outcome. So you need to measure size measure and effort vs. other parameters and keep on fine-tuning your estimation process. There are some LOC-to-effort measures available in the literature, but they are not very accurate in your domain, using the technology you are using, and the team you have.
Other proxies for problem size are function points and story points. My experience on function points is that they are rarely worth the effort. On the other hand, story points in agile methods work very well since they are deliberately abstract (thus avoiding a lot of problems with with LOC) and measured on a sprint-by-sprint basis, with instant feedback into following sprints.
No, it isn't. The reason is simple: if you produce a new line of code during your development, are you one step closer to a solution? If you estimated 1000 lines of code to complete a task, are you now 0.1% complete with that task?
Lines of code can be used as a metric but only in the negative sense: for a greater number of lines of code, it is reasonable to assume that you have a greater number of bugs. Based on historical data, there is generally a linear correlation between lines of code and bug count.
Here are some useful and measurable factors that are worth considering:
Hours of labor.
Dollars spent: this is a good one because it strongly enforces the reality that you'd rather find bugs at the developer's desktop than in the hands of a tester or customer).
Milestones met: is the system available for the customers on the right date?
Requirements completed: this can be a funny one - what if you discover a new customer need during the project?
In short, lines of code is very nearly the worst possible metric you could ever use.
The only way to get any reasonable estimate on project duration is to COMPLETELY implement and deliver some subset of the final requirements. Then you can estimate the remaining requirements by comparing their complexity against the completed work.
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I want to know how to perform "spectral change detection" for the classification of vocal & non-vocal segments of a song. We need to find the spectral changes from a spectrogram. Any elaborate information about this, particularly involving MATLAB?
Separating out distinct signals from audio is a very active area of research, and it is a very hard problem. This is often called Blind Signal Separation in the literature. (There is some MATLAB demo code in the previous link.
Of course, if you know that there is vocal in the music, you can use one of the many vocal separation algorithms.
As others have noted, solving this problem using only raw spectrum analysis is a dauntingly hard problem, and you're unlikely to find a good solution to it. At best, you might be able to extract some of the vocals and a few extra crossover frequencies from the mix.
However, if you can be more specific about the nature of the audio material you are working with here, you might be able to get a little bit further.
In the worst case, your material would be normal mp3's of regular songs -- ie, a full band + vocalist. I have a feeling that this is the case you are probably looking at given the nature of your question.
In the best case, you have access to the multitrack studio recordings and have at least a full mixdown and an instrumental track, in which case you could extract the vocal frequencies from the mix. You would do this by generating an impulse response from one of the tracks and applying it to the other.
In the middle case, you are dealing with simple music which you could apply some sort of algorithm tuned to the parameters of the music to. For instance, if you are dealing with electronic music, you can use to your advantage the stereo width of the track to eliminate all mono elements (ie, basslines + kicks) to extract the vocals + other panned instruments, and then apply some type of filtering and spectrum analysis from there.
In short, if you are planning on making an all-purpose algorithm to generate clean acapella cuts from arbitrary source material, you're probably biting off more than you can chew here. If you can specifically limit your source material, then you have a number of algorithms at your disposal depending on the nature of those sources.
This is hard. If you can do this reliably you will be an accomplished computer scientist. The most promising method I read about used the lyrics to generate a voice only track for comparison. Again, if you can do this and write a paper about it you will be famous (amongst computer scientists). Plus you could make a lot of money by automatically generating timings for karaoke.
If you just want to decide wether a block of music is clean a-capella or with instrumental background, you could probably do that by comparing the bandwidth of the signal to a normal human singer bandwidth. Also, you could check for the base frequency, which can only be in a pretty limited frequency range for human voices.
Still, it probably won't be easy. However, hearing aids do this all the time, so it is clearly doable. (Though they typically search for speech, not singing)
first sync the instrumental with the original, make sure they are the same length and bitrate and start and end at the exact time and convert them to .wav
then do something like
I = wavread(instrumental.wav);
N = wavread(normal.wav);
i = inv(I);
A = (N - i); // it could be A = (N * i) or A = (N + i) you'll have to play around
wavwrite(A, acapella.wav)
that should do it.. a little linear algebra goes a long way.