I would like to change the volume of a music track but only for specific time ranges.
For example, I would like to apply a gain of -6dB between 0s to 1s ; and between 5s to 6s and leave the rest of the soundtrack untouched.
I have seen that I can access the data of the signal with
AudioSegment(…).get_array_of_samples()
I tried to convert it to an numpy.array in order to divide it with a float (100.0) and getting it back with AudioSegment(…)._spawn() but the result is not meeting my expectations... (I even felt it increased the volume)
If required, I can provide a MWE but before, I wanted to verify my idea with some experienced people.
Best regards
The proposed approach is finally valid.
I have found that I had a mistake in the slices corresponding to [0,1] seconds and [5,6] seconds.
Fixing the slices made the processing valid.
Related
We graph fast counters with sum(rate(my_counter_total[1m])) or with sum(irate(my_counter_total[20s])). Where the second one is preferrable if you can always expect changes within the last couple of seconds.
But how do you graph slow counters where you only have some increments every couple of minutes or even hours? Having values like 0.0013232/s is not very human friendly.
Let's say I want to graph how many users sign up to our service (we expect a couple of signups per hour). What's a reasonable query?
We currently use the following to graph that in grafana:
Query: 3600 * sum(rate(signup_total[1h]))
Step: 3600s
Resolution: 1/1
Is this reasonable?
I'm still trying to understand how all those parameters play together to draw a graph. Can someone explain how the range selector ([10m]), the rate() and the irate() functions, the Step and Resolution settings in grafana influence each other?
That's a correct way to do it. You can also use increase() which is syntactic sugar for using rate() that way.
Can someone explain how the range selector
This is only used by Prometheus, and indicates what data to work over.
the Step and Resolution settings in grafana influence each other?
This is used on the Grafana side, it affects how many time slices it'll request from Prometheus.
These settings do not directly influence each other. However the resolution should work out to be smaller than the range, or you'll be undersampling and miss information.
The 3600 * sum(rate(signup_total[1h])) can be substituted with sum(increase(signup_total[1h])) . The increase(counter[d]) function returns counter increase on the given lookbehind window d. E.g. increase(signup_total[1h]) returns the number of signups during the last hour.
Note that the returned value from increase(signup_total[1h]) may be fractional even if signup_total contains only integer values. This is because of extrapolation - see this issue for technical details. There are the following solutions for this issue:
To use offset modifier: signup_total - (signup_total offset 1h) . This query returns correct results if signup_total wasn't reset to zero during the last hour. In this case the sum(signup_total - (signup_total offset 1h)) is roughly equivalent to sum(increase(signup_total[1h])), but returns more accurate integer results.
To use VictoriaMetrics. It returns the expected integer results from increase() out of the box. See this article and this comment for technical details.
I'm trying to make a program run as accurately as possible while staying at a fixed frame rate. How do you do this?
Formally, I have some parameter b in [0,1] that I can set to determine how accurate my computations are (where 0 is least accurate, 0.5 is fairly accurate, and 1 is very accurate). The higher this is, the lower frame rate I will get.
However, there is a "lag", where after changing this parameter, the frame rate won't change until d milliseconds afterwards, where d can vary and is unknown.
Is there a way to change this parameter in a way that prevents "wiggling"? The problem is that if I am experiencing a low frame rate, if I increase the parameter then measure again, it will only be slightly higher, so I will need to increase it more, and then the framerate will be too slow, so I need to decrease the parameter, and I get this oscillating behavior. Is there a way to prevent this? I need to be as reactive as possible in doing this, because changing too slowly will cause the framerate to be incorrect for too long.
Looks like you need an adaptive feedback dampener. Trying an electrical circuit analogy :)
I'd first try to get more info about how the circuit's input signal and responsiveness look like. So I'd first make the algorithm update b not with the desired values but with the previous values plus or minus (as needed towards the desired value) a small fixed increment, say .01 instead (ignore the sloppy response time for now). While doing so I'd collect and plot/analyze the "desired" b values, looking for:
the general shape of the changes: smooth or rather "steppy" or "spiky"? (spiky would require a stronger dampening to prevent oscillations, steppy would require a weaker dampening to prevent lagging)
the maximum/typical/minimum changes in values from sample to sample
the distribution of the changes in values from sample to sample (I'd plan the algorithm to react best for changes in a typical range, say 20-80% range and consider acceptable lagging for changes higher than that or oscillations for values lower than that)
The end goal is to be able to obtain parameters for operating alternatively in 2 modes:
a high-speed tracking mode (also the system's initial mode)
a normal tracking mode
In high-speed tracking mode the b value updates can be either:
not dampened - the update value is the full desired value - only if the changes shape is not spiky and only in the 1st b update after entering the high-speed tracking mode. This would help reduce lagging.
dampened - the update delta is just a fraction (dampening factor) of the desired delta and reflects the fact that the effect of the previous b value update might not be completely reflected in the current frame rate due to d. Dampening helps preventing oscillations at the expense of potentially increasing lag (always conflicting requirements). The dampening factor would be higher for a smooth shape and smaller for a spiky shape.
Switching from high-speed tracking mode to normal tracking mode can be done when the delta between b's previous value and its desired value falls below a certain mode change threshold value (eventually maintained for a minimum number of consecutive samples). The mode change threshold value would be initially estimated from the info collected above and/or empirically determined.
In normal tracking mode the delta between b's previous value and its desired value remain below the mode change threshold value and is either ignored (no b update) or and update is made either with the desired value or some average one - tiny course corrections, keeping the frame rate practically constant, no dampening, no lagging, no oscillations.
When in normal tracking mode the delta between b's previous value and its desired value goes above the mode change threshold value the system switches again to the high-speed tracking mode.
I would also try to get a general idea about how the d response time looks like. To do that I'd change the algorithm to only update b with the desired values not at every iteration, but every n iterations apart (maybe even re-try for several n values). This should indicate how many sample periods would generally a b value change take to become fully effective and should be reflected in the dampening factor: the longer it takes for a change to take effect the stronger the dampening should be to prevent oscillations.
Of course, this is just the general idea, a lot of experimental trial/adjustment iterations may be required to reach a satisfactory solution.
and thanks for reading my thread.
I have read some of the previous posts on formatting/normalising input data for a Neural Network, but cannot find something that addresses my queries specifically. I apologise for the long post.
I am attempting to build a radial basis function network for analysing horse racing data. I realise that this has been done before, but the data that I have is "special" and I have a keen interest in racing/sportsbetting/programming so would like to give it a shot!
Whilst I think I understand the principles for the RBFN itself, I am having some trouble understanding the normalisation/formatting/scaling of the input data so that it is presented in a "sensible manner" for the network, and I am not sure how I should formulate the output target values.
For example, in my data I look at the "Class change", which compares the class of race that the horse is running in now compared to the race before, and can have a value between -5 and +5. I expect that I need to rescale these to between -1 and +1 (right?!), but I have noticed that many more runners have a class change of 1, 0 or -1 than any other value, so I am worried about "over-representation". It is not possible to gather more data for the higher/lower class changes because thats just 'the way the data comes'. Would it be best to use the data as-is after scaling, or should I trim extreme values, or something else?
Similarly, there are "continuous" inputs - like the "Days Since Last Run". It can have a value between 1 and about 1000, but values in the range of 10-40 vastly dominate. I was going to scale these values to be between 0 and 1, but even if I trim the most extreme values before scaling, I am still going to have a huge representation of a certain range - is this going to cause me an issue? How are problems like this usually dealt with?
Finally, I am having trouble understanding how to present the "target" values for training to the network. My existing results data has the "win/lose" (0 or 1?) and the odds at which the runner won or lost. If I just use the "win/lose", it treats all wins and loses the same when really they're not - I would be quite happy with a network that ignored all the small winners but was highly profitable from picking 10-1 shots. Similarly, a network could be forgiven for "losing" on a 20-1 shot but losing a bet at 2/5 would be a bad loss. I considered making the results (+1 * odds) for a winner and (-1 / odds) for a loser to capture the issue above, but this will mean that my results are not a continuous function as there will be a "discontinuity" between short price winners and short price losers.
Should I have two outputs to cover this - one for bet/no bet, and another for "stake"?
I am sorry for the flood of questions and the long post, but this would really help me set off on the right track.
Thank you for any help anyone can offer me!
Kind regards,
Paul
The documentation that came with your RBFN is a good starting point to answer some of these questions.
Trimming data aka "clamping" or "winsorizing" is something I use for similar data. For example "days since last run" for a horse could be anything from just one day to several years but tends to centre in the region of 20 to 30 days. Some experts use a figure of say 63 days to indicate a "spell" so you could have an indicator variable like "> 63 =1 else 0" for example. One clue is to look at outliers say the upper or lower 5% of any variable and clamp these.
If you use odds/dividends anywhere make sure you use the probabilities ie 1/(odds+1) and a useful idea is to normalize these to 100%.
The odds or parimutual prices tend to swamp other predictors so one technique is to develop separate models, one for the market variables (the market model) and another for the non-market variables (often called the "fundamental" model).
In the second line of the program’s output, notice that the value of
331.79, which is assigned to floatingVar, is actually displayed as 331.790009.The reason for this inaccuracy is the particular way in which numbers are internally represented inside the computer.You
have probably come across the same type of inaccuracy when dealing
with numbers on your calculator. If you divide 1 by 3 on your
calculator, you get the result .33333333, with perhaps some additional
3s tacked on at the end.The string of 3s is the calculator’s
approximation to one third.Theoretically, there should be an infinite
number of 3s. But the calculator can hold only so many digits, thus
the inherent inaccuracy of the machine.The same type of inaccuracy
applies here: Certain floatingpoint values cannot be exactly
represented inside the computer’s memory.
the above quote comes from Programming in Objective-C – 4th edition
And this post answered a little part but not the kind of answer i'm trying to look for.
Will try to find another book about this later in the day.
Anyway if anyone would like to answer this question, thanks!
I found this on an "interview questions" site and have been pondering it for a couple of days. I will keep churning, but am interested what you guys think
"10 Gbytes of 32-bit numbers on a magnetic tape, all there from 0 to 10G in random order. You have 64 32 bit words of memory available: design an algorithm to check that each number from 0 to 10G occurs once and only once on the tape, with minimum passes of the tape by a read head connected to your algorithm."
32-bit numbers can take 4G = 2^32 different values. There are 2.5*2^32 numbers on tape total. So after 2^32 count one of numbers will repeat 100%. If there were <= 2^32 numbers on tape then it was possible that there are two different cases – when all numbers are different or when at least one repeats.
It's a trick question, as Michael Anderson and I have figured out. You can't store 10G 32b numbers on a 10G tape. The interviewer (a) is messing with you and (b) is trying to find out how much you think about a problem before you start solving it.
The utterly naive algorithm, which takes as many passes as there are numbers to check, would be to walk through and verify that the lowest number is there. Then do it again checking that the next lowest is there. And so on.
This requires one word of storage to keep track of where you are - you could cut down the number of passes by a factor of 64 by using all 64 words to keep track of where you're up to in several different locations in the search space - checking all of your current ones on each pass. Still O(n) passes, of course.
You could probably cut it down even more by using portions of the words - given that your search space for each segment is smaller, you won't need to keep track of the full 32-bit range.
Perform an in-place mergesort or quicksort, using tape for storage? Then iterate through the numbers in sequence, tracking to see that each number = previous+1.
Requires cleverly implemented sort, and is fairly slow, but achieves the goal I believe.
Edit: oh bugger, it's never specified you can write.
Here's a second approach: scan through trying to build up to 30-ish ranges of contiginous numbers. IE 1,2,3,4,5 would be one range, 8,9,10,11,12 would be another, etc. If ranges overlap with existing, then they are merged. I think you only need to make a limited number of passes to either get the complete range or prove there are gaps... much less than just scanning through in blocks of a couple thousand to see if all digits are present.
It'll take me a bit to prove or disprove the limits for this though.
Do 2 reduces on the numbers, a sum and a bitwise XOR.
The sum should be (10G + 1) * 10G / 2
The XOR should be ... something
It looks like there is a catch in the question that no one has talked about so far; the interviewer has only asked the interviewee to write a program that CHECKS
(i) if each number that makes up the 10G is present once and only once--- what should the interviewee do if the numbers in the given list are present multple times? should he assume that he should stop execting the programme and throw exception or should he assume that he should correct the mistake by removing the repeating number and replace it with another (this may actually be a costly excercise as this involves complete reshuffle of the number set)? correcting this is required to perform the second step in the question, i.e. to verify that the data is stored in the best possible way that it requires least possible passes.
(ii) When the interviewee was asked to only check if the 10G weight data set of numbers are stored in such a way that they require least paases to access any of those numbers;
what should the interviewee do? should he stop and throw exception the moment he finds an issue in the algorithm they were stored in, or correct the mistake and continue till all the elements are sorted in the order of least possible passes?
If the intension of the interviewer is to ask the interviewee to write an algorithm that finds the best combinaton of numbers that can be stored in 10GB, given 64 32 Bit registers; and also to write an algorithm to save these chosen set of numbers in the best possible way that require least number of passes to access each; he should have asked this directly, woudn't he?
I suppose the intension of the interviewer may be to only see how the interviewee is approaching the problem rather than to actually extract a working solution from the interviewee; wold any buy this notion?
Regards,
Samba