Let's say I have a online game where users can play each other. I have list of players, all players categorized as experts, average, below average. A new player wants to play this game, and the system choose expert level player against new player. what would be the winning probability of the new player?
ex: expert player data looks as:
total games: 45,
win: 30,
draw: 10,
loss: 5
Is it possible to do probability between new player vs expert player? If yes, what statistics can be considered?
Thanks in advance
I would take a subset of the data which corresponds to expert vs. average games
If the expert in question has enough data (do a binomial test to check if his win rate is statistically significant), then you can rely on his/her data alone. Eg If this particular expert has played 100 games against average players and won 80, that's significant, and you can say he/she has a 80% change of winning
If the expert doesn't have enough data (ie to be significant), you can combine data from ALL expert vs. average games to compensate, although this raises another question: how did this player get awarded the expert rank if we believe there aren't enough games to determine this?
I believe the main issue here is that you reduce all possible skill levels to 3 (expert vs. average vs. below average), whereas in reality there might be more granularity to how good or bad people are, thus your model might be over simplistic. Introducing more levels would help address that issue (eg 5 levels doesn't seem like too much for people to grasp and might already perform better than 3). Alternatively you can also try to calculate probabilities based on more detailed properties (eg win rate, # days user has been playing, age, gender...) even if it's something you only keep to yourself (ie players only see the n ranking levels but you have more detailed properties to do your calculations).
Related
I am trying to create an Agent in Rust that uses a scoring function to determine the best move on a 2D uniform cost grid. The specifics of the game aren't very relevant, other than knowing that each turn you can choose to make one of 4 moves (up, down, left or right) and you are competing against other AIs who are playing on the same board. Currently the AI makes "branches" of possible paths it could make into the future using several different simple algorithms such as using A* to find enemies or food. Several characteristics are saved as the future simulations run including the number of enemies we killed on that branch, amount of food we ate and how long the future branch lasted before we died.
Once we are ready to make our move, we give each future predicting branch a score and go in the direction with the highest average score. This score is essentially a sum of each characteristic mentioned previously multiplied by a constant. For example the score may be 30 * number of food eaten + 100 * number of enemies killed. However, the number 30 and 100 were chosen almost at random through experimentation. If the snake died from not eating food then I increase the score multiplier for eating food for example. However, there are 10 different characteristics each with their own weight. Figuring out the relationship between them all manually is both time consuming and doesn't easily converge onto the optimal strategy.
Here in lies my issue. I would like to find a way to "train" the values for the AI through a process sort of like Q-Learning. There is a very clear terminal condition when you win or lose which helps. My currently idea is creating a table with 100 possible values of each parameter, then play 100 games with each combination and record the win rate. However, this would take (1000 choose 10) * 100 games or 2.6E25 games. It seems like there should be a smarted way to eliminate bad combinations using some form of loss minimization. If anybody has suggestions on tuning these parameters without a neural network, it would be greatly appreciated.
Suppose you have a set of transcribed customer service calls between customers and human agents, where on average each call's length is 7 minutes. Customers will mostly call because of issues they have with the product. Let's assume that a human can assign one label per axis per call:
Axis 1: What was the problem from the customer's perspective?
Axis 2: What was the problem from the agent's perspective?
Axis 3: Could the agent resolve the customer's issue?
Based on the manually labeled texts you want to train a text classifier that shall predict a label for each call for each of the three axes. But the labeling of recordings takes time and costs money. On the other hand you need a certain amount of training data to get good prediction results.
Given the above assumptions, how many manually labeled training texts would you start with? And how do you know that you need more labeled training texts?
Maybe you've worked on a similar task before and can give some advice.
UPDATE (2018-01-19): There's no right or wrong answer to my question. Ok, ideally, somebody worked on exactly the same task, but that's very unlikely. I'll leave the question open for one more week and then accept the best answer.
This would be tricky to answer but I will try my best based on my experience.
In the past, I have performed text classification on 3 datasets; the number in the bracket indicates how big my dataset was: restaurant reviews (50K sentences), reddit comments (250k sentences) and developer comments from issue tracking systems (10k sentences). Each of them had multiple labels as well.
In each of the three cases, including the one with 10k sentences, I achieved an F1 score of more than 80%. I am stressing on this dataset specifically because I was told by some that the size is less for this dataset.
So, in your case, assuming you have atleast 1000 instances (calls that include conversation between customer and agent) of average 7 minute calls, this should be a decent start. If the results are not satisfying, you have the following options:
1) Use different models (MNB, Random Forest, Decision Tree, and so on in addition to whatever you are using)
2) If point 1 gives more or less similar results, check the ratio of instances of all the classes you have (the 3 axis you are talking about here). If they do not share a good ratio, get more data or try out the different balancing techniques if you cannot get more data.
3) Another way would be to classify them at a sentence level than message or conversation level to generate more data and individual labels for sentences rather than message or the conversation itself.
I want to calculate room noise level with the computer's microphone. I record noise as an audio file, but how can I calculate the noise dB level?
I don't know how to start!
All the previous answers are correct if you want a technically accurate or scientifically valuable answer. But if you just want a general estimation of comparative loudness, like if you want to check whether the dog is barking or whether a baby is crying and you want to specify the threshold in dB, then it's a relatively simple calculation.
Many wave-file editors have a vertical scale in decibels. There is no calibration or reference measurements, just a simple calculation:
dB = 20 * log10(amplitude)
The amplitude in this case is expressed as a number between 0 and 1, where 1 represents the maximum amplitude in the sound file. For example, if you have a 16 bit sound file, the amplitude can go as high as 32767. So you just divide the sample by 32767. (We work with absolute values, positive numbers only.) So if you have a wave that peaks at 14731, then:
amplitude = 14731 / 32767
= 0.44
dB = 20 * log10(0.44)
= -7.13
But there are very important things to consider, specifically the answers given by the others.
1) As Jörg W Mittag says, dB is a relative measurement. Since we don't have calibrations and references, this measurement is only relative to itself. And by that I mean that you will be able to see that the sound in the sound file at this point is 3 dB louder than at that point, or that this spike is 5 decibels louder than the background. But you cannot know how loud it is in real life, not without the calibrations that the others are referring to.
2) This was also mentioned by PaulR and user545125: Because you're evaluating according to a recorded sound, you are only measuring the sound at the specific location where the microphone is, biased to the direction the microphone is pointing, and filtered by the frequency response of your hardware. A few feet away, a human listening with human ears will get a totally different sound level and different frequencies.
3) Without calibrated hardware, you cannot say that the sound is 60dB or 89dB or whatever. All that this calculation can give you is how the peaks in the sound file compares to other peaks in the same sound file.
If this is all you want, then it's fine, but if you want to do something serious, like determine whether the noise level in a factory is safe for workers, then listen to Paul, user545125 and Jörg.
You do need reference hardware (i.e., a reference mic) to calculate noise level (dB SPL, or sound pressure level). One thing Radio Shack sells is a $50 dB SPL meter. If you're doing scientific calculations, I wouldn't use it. But if the goal is to get a general idea of a weighted measurement (dBA or dBC) of the sound pressure in a given environment, then it might be useful. As a sound engineer, I use mine all the time to see how much sound volume I'm generating while I mix. It's usually accurate to within 2 dB.
That's my answer. The rest is FYI stuff.
Jorg is correct that dB SPL is a relative measurement. All decibel measurements are. But you've implied a reference of 0 dB SPL, or 20 micropascals, scientifically agreed to be the most quiet sound a human ear can detect (though, understandably, what a person can actually hear is very difficult to determine). This, according to Wikipedia, is about the sound of a flying mosquito from about 10 feet away (http://en.wikipedia.org/wiki/Decibel).
By assuming you don't understand decibels, I think Jorg is just trying to out-geek you. He clearly didn't give you a practical answer. :-)
Unweighted measurements (dB, instead of dBA or dBC) are rarely used, because most sound pressure is not detected by the human ear. In a given office environment, there is usually 80-100 dB SPL (sound pressure level). To give you an idea of exactly how much is not heard, in the U.S., occupational regulations limit noise exposure to 80 dBA for a given 8-hour work shift (80 dBA is about the background noise level of your average downtown street - difficult, but not impossible to talk over). 85 dBA is oppressive, and at 90, most people are trying to get away. So the difference between 80 dB and 80 dBA is very significant -- 80 dBA is difficult to talk over, and 80 dB is quite peaceful. :-)
So what is 'A' weighting? 'A' weighting compensates for the fact that we don't perceive lower frequency sounds as well as high frequency sounds (we hear 20 Hz to 20,000 Hz). There's a lot of low-end rumble that our ears/brains pretty much ignore. In addition, we're more sensitive to a certain midrange (1000 Hz to 4000 Hz). Most agree that this frequency range contains the sounds of consonants of speech (vowels happen at a much lower frequency). Imagine talking with just vowels. You can't understand anything. Thus, the ability of a human to be able to communicate (conventionally) rests in the 1kHz-5kHz bump in hearing sensitivity. Interestingly, this is why most telephone systems only transmit 300 Hz to 3000 Hz. It was determined that this was the minimal response needed to understand the voice on the other end.
But I think that's more than you wanted to know. Hope it helps. :-)
You can't easily measure absolute dB SPL, since your microphone and analogue hardware are not calibrated. You may be able to do an approximate calibration for a particular hardware set up but you would need to repeat this for every different microphone and hardware set up that you plan to support.
If you do have some kind of SPL reference source that you can use then then it gets easier:
use your reference source to generate a tone at a known dB SPL - measure this
measure the ambient noise
calculate noise level = 20 * log10 (V_noise / V_ref) + dB_ref
Of course this assumes that the frequency response of your microphone and audio hardware is reasonably flat and that you just want a flat (unweighted) noise figure. If you want a weighted (e.g. A-weight) noise figure then you'll have to do rather more processing.
According to Merchant et al. (section 3.2 in the appendix: "Measuring acoustic habitats", Methods in Ecology and Evolution, 2015), you can actually calculate absolute, calibrated SPL values using manufacturer specifications by subtracting a correction term S to your relative (scaled to maximum) SPL values:
S = M + G + 20*log10(1/Vadc) + 20*log10(2^Nbit-1)
where M is the sensitivity of the transducer (microphone) re 1 V/Pa. G is the gain applied by the user. Vadc is the zero-to-peak voltage, given by multiplying the rms ADC voltage by a conversion factor of squareroot(2). Nbit is the bit sampling depth.
The last term is necessary if your system scales the amplitude by its maximum.
The correction will be more accurate using end-to-end calibration with sound calibrators.
Note that the formula above is dependent on frequency, but you could apply it over a wider frequency range if your microphone has a flat frequency response.
You can't. dB is a relative unit, IOW it is a unit for comparing two measurements against each other. You can only say that measurement A is x dB louder than measurement B, but in your case you only have one measurement. Therefore, it simply isn't possible to calculate the dB level.
The short answer is: you cannot do sound level measurements with your laptop, nor with your cellphone, etc., for all the reasons outlined previously, plus the fact your cellphone, laptop, etc. use compression algorithms to assure that everything recorded is within the hardware capability. So, if for example you measure a sound then run it through signal processing software such as Head Artemis or LMS Test.Lab, the indicated sound pressure level will always be in the neighborhood of 80 dB(A) regardless of the true level. I can say this from having used cellphone or laptop audio to get an idea of a noise frequency spectrum, while taking level measurements using a calibrated sound level meter. Interestingly, Radio Shack used to sell a microphone intended for speech input while videoconferencing that had very flat frequency response over a broad range, and only cost about $15.
I use a sound level calibrator.
It produces 94 dB or 114dB at 1 KHz
wich is a frecuency where weighting
filters share the same level.
With calibrator at 114dB I adjust mic gain to reach almost full scale
input simply watching a sound card based virtual osciloscope.
Now I know Vref # 114dB.
I developed a simple software based SPL meter
that can be provided if needed. You can use REW too.
You hace to know that PC hardware hardly
reaches 60 dB of dynamic range so calibrating
#114 dB it wont read less than 54dB, wich
is pretty high if you consider that sleeping
is good with less than 35 dB A.
In this case you can calibrate at 94dB
and then you may measure down to 34dB
but again you will hit pc and mic self noise
wich may you prevent to reach such low levels.
Anyway, once calibrated, measures at 114dB
and 94dB should read fine.
Note: the lab standard pistonphone calibrator operates at 250 Hz.
Well! I Used RobertT's Method But It Always Giving Me Oveflow Exception, Then I Used:- int dB = -36 - (value * -1), The Exception Gone, I Don't Know Whether It's Telling dB Values, If You Knew Using Code Given Below, Please Comment Me Whether it's A dB Value or not.
VB.NET:-
Dim dB As Integer = -36 - (9 * -1)
C#:-
int dB = -36 - (9 * -1)
As a follow-up to my previous question, if I want my smartphone application to detect a certain musical note, and I only need to know whether the incoming sound is that musical note or not, with a certain amount of fuzziness, to allow the note to be off-key by x cents.
Given that, is there a superior method over others for speed and accuracy? That is, by knowing that the note you are looking for is, say, a #C3, how best to tell if that note is present or not? I'm assuming that looking for a single note would be easier than separating out all waveforms, and then looking at the results for the fundamental frequency.
In the responses to my original question, one respondent suggested that autocorrelation might work well if you know that the notes are within a certain range. I wonder if autocorrelation would then work even better, if you only have to check for the presence or absence of a certain note (+/- x cents).
Those methods being:
Kiss FFT
FFTW
Discrete Wavelet Transform
autocorrelation
zero crossing analysis
octave-spaced filters
DWT
Any thoughts would be appreciated.
As you describe it, you just need to determine if a particular pitch is present. A very simple (fast) detector would just record the equivalent of one period of the waveform, then record another period and correlate them, like an oversimplified (single-lag) autocorrelation. If there's a high match, you know the waveform being recorded is repeating at around the same period, or a harmonic of it.
For instance, to detect 1 kHz, record 1 ms of audio (48 samples at 48 kHz), then record another 1 ms, and compare them (correlate = multiply all samples and sum). If they line up (correlation above some threshold), then you're listening to 1 kHz, 2 kHz, 3 kHz, or some other multiple. Doing several periods would give you more confidence on the match.
A true autocorrelation would tell you which harmonic, specifically, if that's important to you.
I have a bunch of different audio recordings in WAV format (all different instruments and pitches), and I want to "normalize" them so that they all sound approximately the same volume when played.
I've tried measuring the average sample magnitude (the sum of all absolute values divided by the number of samples), but normalizing by this measurement doesn't work very well. I think this method isn't working because it doesn't take into account the frequency of the sounds, and I know that higher-frequency recordings sound louder than lower-frequency sounds of the same amplitude.
Does anyone know a good method for measuring the loudness of a sound?
Root Mean Square is often used to estimate the loudness of sound files. This is because a sound that is very loud might not be perceived that way if it is very short. Also remember that power increases exponentially with the square of amplitude.
The audio geeks at Hydrogen Audio know a ton about this stuff...check out their free Replay Gain software. You may not need to do any programming at all.
EDIT: Included comment feedback on power vs. amplitude.
To add to PeterAllenWebb's response:
Before you calculate the RMS, you should "center" your sample first (think of a 5-minute .wav where each sample has the maximum +amplitude). The best way to do that is to use a highpass filter at a subsonic frequency.
That would still not take the frequencies that humans are sensitive to in count. To do that, you could use A-weighting. There's a page where you can calculate it online:
http://www.diracdelta.co.uk/science/source/a/w/aweighting/source.html
The code seems to be here:
http://www.diracdelta.co.uk/science/source/a/w/aweighting/multicalc.js
Well not being an expert on audio and adding to the previous comment, you should figure out what you define as the "shortest amount of time for peak power" and then just convert the wave to raw floating point and use RMS over the stretch of time and continuously take chunks of that length of time, find the MAX and there you have your highest peak power.
To reiterate what some other people have said, use RMS value to estimate the "loudness" of a passage of sound.
But, if you're dealing with impulsive sounds like plucking or drum hits, you'd want to do a sliding RMS value and pick out only the peak RMS value. Measure 100 ms of the sound, slide the window, measure again, etc. and then normalize according to the largest value you find.
Definitely remove any DC value before doing the RMS, and A-weighting will make it more like how we hear. Here's code for A-weighting in MATLAB/Octave and Python.
I might be way off here, but, if you have wavepad you can load in multiple files and mess with the volumes a little bit so they are all the same. Also, if you have certain sections of a file that are louder, you can select that section and lower the volume for that one section.
EDIT: And sorry, it;s not really a "method" for measuring volume, but if you just need to make them all the same this should work fine.