I have a big problem I want to implement my neuronal neutwork with 2 neurons outputs. Sth like that :
And I want to use backpropagation algorithm, but I don't know how to calculate a error, because I have a output with 2 neurons, when I have a only one neuron on a output that's very easy to use a backpropagation algorithm from one exit error, but with two neurons? I thinking about calculate error for every output seperately but then I must calculate seperately back propagation for 2 cases and I get "two different hidden layers" (For every neuron in hidden layer I have a weights for two cases). Mayby anyone knows some better solutions?
I will be very gratefull for any help.
Logically thinking, the first layer of weights should give you a representation (the hidden layer) that is useful for predicting both outputs. So, this layer should be updated based on the error made in both outputs. But the next layer of weights are separate for each output node, so should get separate weight updates.
So, on second layer weights, the weight updates will be calculated separately based on the respective outputs. For the first layer of weights, I would first calculate error derivatives backpropagating from each output separately and then simply combine them to get the final error derivative. Then apply learning rate to get the weight updates.
Watch out for the dynamic range of your outputs. For example, if one output is producing some real value of range [0,10] and another is producing values in range [-1000,1000] then your updates will be dominated by the one with larger range. You can
add a preprocessing step that would change your data set to have same dynamic range in both outputs. Also, add a postprocessing step to restore the actual range.
formulate the error functions for each output so that they produce error values of same dynamic range.
Related
I want to build an application that would do something equivalent to running lsof (maybe changing it to output differently, because string processing may mean it is not real time enough) in a loop and then associate each line (entries) with what iteration it was present in, what I will be referring further as frames, as later on it will be better for understanding. My intention with it is that showing the times in which files are open by applications can reveal something about their structure, while not having big impact on their execution, which is often a problem. One problem I have is on processing the output, which would be a table relating "frames X entry", for that I am already anticipating that I will have wildly variable entry lengths. Which can fall in that problem of representing on geometry when you have very different scales, the smaller get infinitely small, while the bigger gets giant and fragmentation makes it even worse; so my question is if plotting libraries deal with this problem and how they do it
The easiest and most well-established technique for showing both small and large values in reasonable detail is a logarithmic scale. Instead of plotting raw values, plot their logarithms. This is notoriously problematic if you can have zero or even negative values, but as I understand your situations all your lengths would be strictly positive so this should work.
Another statistical solution you could apply is to plot ranks instead of raw values. Take all the observed values, and put them in a sorted list. When plotting any single data point, instead of plotting the value itself you look up that value in the list of values (possibly using binary search since it's a sorted list) then plot the index at which you found the value.
This is a monotonous transformation, so small values map to small indices and big values to big indices. On the other hand it completely discards the actual magnitude, only the relative comparisons matter.
If this is too radical, you could consider using it as an ingredient for something more tuneable. You could experiment with a linear combination, i.e. plot
a*x + b*log(x) + c*rank(x)
then tweak a, b and c till the result looks pleasing.
I just started a Machine learning class and we went over Perceptrons. For homework we are supposed to:
"Choose appropriate training and test data sets of two dimensions (plane). Use 10 data points for training and 5 for testing. " Then we are supposed to write a program that will use a perceptron algorithm and output:
a comment on whether the training data points are linearly
separable
a comment on whether the test points are linearly separable
your initial choice of the weights and constants
the final solution equation (decision boundary)
the total number of weight updates that your algorithm made
the total number of iterations made over the training set
the final misclassification error, if any, on the training data and
also on the test data
I have read the first chapter of my book several times and I am still having trouble fully understanding perceptrons.
I understand that you change the weights if a point is misclassified until none are misclassified anymore, I guess what I'm having trouble understanding is
What do I use the test data for and how does that relate to the
training data?
How do I know if a point is misclassified?
How do I go about choosing test points, training points, threshold or a bias?
It's really hard for me to know how to make up one of these without my book providing good examples. As you can tell I am pretty lost, any help would be so much appreciated.
What do I use the test data for and how does that relate to the
training data?
Think about a Perceptron as young child. You want to teach a child how to distinguish apples from oranges. You show it 5 different apples (all red/yellow) and 5 oranges (of different shape) while telling it what it sees at every turn ("this is a an apple. this is an orange). Assuming the child has perfect memory, it will learn to understand what makes an apple an apple and an orange an orange if you show him enough examples. He will eventually start to use meta-features (like shapes) without you actually telling him. This is what a Perceptron does. After you showed him all examples, you start at the beginning, this is called a new epoch.
What happens when you want to test the child's knowledge? You show it something new. A green apple (not just yellow/red), a grapefruit, maybe a watermelon. Why not show the child the exact same data as before during training? Because the child has perfect memory, it will only tell you what you told him. You won't see how good it generalizes from known to unseen data unless you have different training data that you never showed him during training. If the child has a horrible performance on the test data but a 100% performance on the training data, you will know that he has learned nothing - it's simply repeating what he has been told during training - you trained him too long, he only memorized your examples without understanding what makes an apple an apple because you gave him too many details - this is called overfitting. To prevent your Perceptron from only (!) recognizing training data you'll have to stop training at a reasonable time and find a good balance between the size of the training and testing set.
How do I know if a point is misclassified?
If it's different from what it should be. Let's say an apple has class 0 and an orange has 1 (here you should start reading into Single/MultiLayer Perceptrons and how Neural Networks of multiple Perceptrons work). The network will take your input. How it's coded is irrelevant for this, let's say input is a string "apple". Your training set then is {(apple1,0), (apple2,0), (apple3,0), (orange1,1), (orange2,1).....}. Since you know the class beforehand, the network will either output 1 or 0 for the input "apple1". If it outputs 1, you perform (targetValue-actualValue) = (1-0) = 1. 1 in this case means that the network gives a wrong output. Compare this to the delta rule and you will understand that this small equation is part of the larger update equation. In case you get a 1 you will perform a weight update. If target and actual value are the same, you will always get a 0 and you know that the network didn't misclassify.
How do I go about choosing test points, training points, threshold or
a bias?
Practically the bias and threshold isn't "chosen" per se. The bias is trained like any other unit using a simple "trick", namely using the bias as an additional input unit with value 1 - this means the actual bias value is encoded in this additional unit's weight and the algorithm we use will make sure it learns the bias for us automatically.
Depending on your activation function, the threshold is predetermined. For a simple perceptron, the classification will occur as follows:
Since we use a binary output (between 0 and 1), it's a good start to put the threshold at 0.5 since that's exactly the middle of the range [0,1].
Now to your last question about choosing training and test points: This is quite difficult, you do that by experience. Where you're at, you start off by implementing simple logical functions like AND, OR, XOR etc. There's it's trivial. You put everything in your training set and test with the same values as your training set (since for x XOR y etc. there are only 4 possible inputs 00, 10, 01, 11). For complex data like images, audio etc. you'll have to try and tweak your data and features until you feel like the network can work with it as good as you want it to.
What do I use the test data for and how does that relate to the training data?
Usually, to asses how well a particular algorithm performs, one first trains it and then uses different data to test how well it does on data it has never seen before.
How do I know if a point is misclassified?
Your training data has labels, which means that for each point in the training set, you know what class it belongs to.
How do I go about choosing test points, training points, threshold or a bias?
For simple problems, you usually take all the training data and split it around 80/20. You train on the 80% and test against the remaining 20%.
I have a long time series with some repeating and similar looking signals in it (not entirely periodical). The length of the time series is about 60000 samples. To identify the signals, I take out one of them, having a length of around 1000 samples and move it along my timeseries data sample by sample, and compute cross-correlation coefficient (in Matlab: corrcoef). If this value is above some threshold, then there is a match.
But this is excruciatingly slow (using 'for loop' to move the window).
Is there a way to speed this up, or maybe there is already some mechanism in Matlab for this ?
Many thanks
Edited: added information, regarding using 'xcorr' instead:
If I use 'xcorr', or at least the way I have used it, I get the wrong picture. Looking at the data (first plot), there are two types of repeating signals. One marked by red rectangles, whereas the other and having much larger amplitudes (this is coherent noise) is marked by a black rectangle. I am interested in the first type. Second plot shows the signal I am looking for, blown up.
If I use 'xcorr', I get the third plot. As you see, 'xcorr' gives me the wrong signal (there is in fact high cross correlation between my signal and coherent noise).
But using "'corrcoef' and moving the window, I get the last plot which is the correct one.
There maybe a problem of normalization when using 'xcorr', but I don't know.
I can think of two ways to speed things up.
1) make your template 1024 elements long. Suddenly, correlation can be done using FFT, which is significantly faster than DFT or element-by-element multiplication for every position.
2) Ask yourself what it is about your template shape that you really care about. Do you really need the very high frequencies, or are you really after lower frequencies? If you could re-sample your template and signal so it no longer contains any frequencies you don't care about, it will make the processing very significantly faster. Steps to take would include
determine the highest frequency you care about
filter your data so higher frequencies are blocked
resample the resulting data at a lower sampling frequency
Now combine that with a template whose size is a power of 2
You might find this link interesting reading.
Let us know if any of the above helps!
Your problem seems like a textbook example of cross-correlation. Therefore, there's no good reason using any solution other than xcorr. A few technical comments:
xcorr assumes that the mean was removed from the two cross-correlated signals. Furthermore, by default it does not scale the signals' standard deviations. Both of these issues can be solved by z-scoring your two signals: c=xcorr(zscore(longSig,1),zscore(shortSig,1)); c=c/n; where n is the length of the shorter signal should produce results equivalent with your sliding window method.
xcorr's output is ordered according to lags, which can obtained as in a second output argument ([c,lags]=xcorr(..). Always plot xcorr results by plot(lags,c). I recommend trying a synthetic signal to verify that you understand how to interpret this chart.
xcorr's implementation already uses Discere Fourier Transform, so unless you have unusual conditions it will be a waste of time to code a frequency-domain cross-correlation again.
Finally, a comment about terminology: Correlating corresponding time points between two signals is plain correlation. That's what corrcoef does (it name stands for correlation coefficient, no 'cross-correlation' there). Cross-correlation is the result of shifting one of the signals and calculating the correlation coefficient for each lag.
I'm working on character recognition (and later fingerprint recognition) using neural networks. I'm getting confused with the sequence of events. I'm training the net with 26 letters. Later I will increase this to include 26 clean letters and 26 noisy letters. If I want to recognize one letter say "A", what is the right way to do this? Here is what I'm doing now.
1) Train network with a 26x100 matrix; each row contains a letter from segmentation of the bmp (10x10).
2) However, for the test targets I use my input matrix for "A". I had 25 rows of zeros after the first row so that my input matrix is the same size as my target matrix.
3) I run perform(net, testTargets,outputs) where outputs are the outputs from the net trained with the 26x100 matrix. testTargets is the matrix for "A".
This doesn't seem right though. Is training supposed by separate from recognizing any character? What I want to happen is as follows.
1) Training the network for an image file that I select (after processing the image into logical arrays).
2) Use this trained network to recognize letter in a different image file.
So train the network to recognize A through Z. Then pick an image, run the network to see what letters are recognized from the picked image.
Okay, so it seems that the question here seems to be more along the lines of "How do I neural networks" I can outline the basic procedure here to try to solidify the idea in your mind, but as far as actually implementing it goes you're on your own. Personally I believe that proprietary languages (MATLAB) are an abomination, but I always appreciate intellectual zeal.
The basic concept of a neural net is that you have a series of nodes in layers with weights that connect them (depending on what you want to do you can either just connect each node to the layer above and beneath, or connect every node, or anywhere in betweeen.). Each node has a "work function" or a probabilistic function that represents the chance that the given node, or neuron will evaluate to "on" or 1.
The general workflow starts from whatever top layer neurons/nodes you've got, initializing them to the values of your data (in your case, you would probably start each of these off as the pixel values in your image, normalized to be binary would be simplest). Each of those nodes would then be multiplied by a weight and fed down towards your second layer, which would be considered a "hidden layer" depending on the sum (either geometric or arithmetic sum, depending on your implementation) which would be used with the work function to determine the state of your hidden layer.
That last point was a little theoretical and hard to follow, so here's an example. Imagine your first row has three nodes ([1,0,1]), and the weights connecting the three of those nodes to the first node in your second layer are something like ([0.5, 2.0, 0.6]). If you're doing an arithmetic sum that means that the weighting on the first node in your "hidden layer" would be
1*0.5 + 0*2.0 + 1*0.6 = 1.1
If you're using a logistic function as your work function (a very common choice, though tanh is also common) this would make the chance of that node evaluating to 1 approximately 75%.
You would probably want your final layer to have 26 nodes, one for each letter, but you could add in more hidden layers to improve your model. You would assume that the letter your model predicted would be the final node with the largest weighting heading in.
After you have that up and running you want to train it though, because you probably just randomly seeded your weights, which makes sense. There are a lot of different methods for this, but I'll generally outline back-propagation which is a very common method of training neural nets. The idea is essentially, since you know which character the image should have been recognized, you compare the result to the one that your model actually predicted. If your model accurately predicted the character you're fine, you can leave the model as is, since it worked. If you predicted an incorrect character you want to go back through your neural net and increment the weights that lead from the pixel nodes you fed in to the ending node that is the character that should have been predicted. You should also decrement the weights that led to the character it incorrectly returned.
Hope that helps, let me know if you have any more questions.
I'm working on a project to recognize simple audio patterns. I have two data sets, each made up of between 4 and 32 note/duration pairs. One set is predefined, the other is from an incoming data stream. The length of the two strongly correlated data sets is often different, but roughly the same "shape". My goal is to come up with some sort of ranking as to how well the two data sets correlate/match.
I have converted the incoming frequencies to pitch and shifted the incoming data stream's pitch so that it's average pitch matches that of the predefined data set. I also stretch/compress the incoming data set's durations to match the overall duration of the predefined set. Here are two graphical examples of data that should be ranked as strongly correlated:
http://s2.postimage.org/FVeG0-ee3c23ecc094a55b15e538c3a0d83dd5.gif
(Sorry, as a new user I couldn't directly post images)
I'm doing this on a 8-bit microcontroller so resources are minimal. Speed is less an issue, a second or two of processing isn't a deal breaker.
It wouldn't surprise me if there is an obvious solution, I've just been staring at the problem too long. Any ideas?
Thanks in advance...
Couldn't see the graphic, but... Divide the spectrum into bins. You've probably already done this already , but they may be too fine. Depending on your application, consider dividing the spectrum into, say 16 or 32 bins, maybe logarithmically, since that is how we hear. Then, compare the ratios of the power in each bin. E.g, compare the ratio of 500 Hz to 1000 Hz in the first sample with that same ratio in the 2nd sample. That gets rid of any problem with unequal amplitudes of the samples.
1D signal matching is often done with using the convolution function. However, this may be processor intensive.
A simpler algorithm that could be used is to first check if the durations of each note the two signals are roughly equal. Then if check the next-frequency pattern of the two signals are the same. What I mean by next-frequency pattern is to decompose the ordered list of frequencies to an ordered list of whether or not the next frequency is higher or lower. So something that goes 500Hz to 1000Hz to 700Hz to 400Hz would simply become Higher-Lower-Lower. This may be good enough, depending on your purposes.