Hi every one i am performing the basic Hecgeo-ras Tutorial when i exported the geometry in Hec-ras and computed the Steady Flow Analysis in hec-ras it gives me this error 'a horizontal manning n value needs to be specified on first station'
Can any one help me about this.
When using HEC-GeoRAS, it is the user's responsibility to assign Manning's n coefficients to each cross section. That is, unless you first created a LandUse feature class in the map document. If you created an empty LandUse feature class then the "N_Value" fields will be blank and since the calculations in the program start at the first station, you will see an error just as you stated.
From the Purdue University Tutorial:
To assign Manning’s n to cross-sections, click on RAS Geometry->Manning’s n
Values->Extract n Values. Confirm LandUse for Land Use, choose N_Value for Manning Field, XSCutLines for XS Cut Lines, leave the default name Manning for XS Manning Table, and click OK.
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Referring to the original paper on CycleGAN i am confused about this line
The optimal G thereby translates the domain X to a domain Yˆ
distributed identically to Y . However, such a translation does not
guarantee that an individual input x and output y are paired up in a
meaningful way – there are infinitely many mappings G that will induce
the same distribution over yˆ.
I understand there are two sets of images and there is no pairing between them so when generator will taken one image lets say x from set X as input and try to translate it to an image similar to the images in Y set then my question is that there are many images present in the set Y so which y will our x be translated into? There are so many options available in set Y. Is that what is pointed out in these lines of the paper that i have written above? And is this the reason we take cyclic loss to overcome this problem and to create some type of pairing between any two random images by converting x to y and then converting y back to x?
The image x won't be translated to a concrete image y but rather to a "style" of the domain Y. The input is fed to the generator, which tries to produce a sample from the desired distribution (the other domain), the generated image then goes to the discriminator, which tries to predict if the sample is from the actual distribution or produced by the generator. This is just the normal GAN workflow.
If I understand it correctly, in the lines you quoted, authors explain the problems that arise with adversarial loss. They say it again here:
Adversarial training can, in theory, learn mappings G and F that produce outputs identically distributed as target domains Y and X respectively. However, with large enough capacity, a network can map the same set of input images to any random permutation of images in the target domain, where any of the learned mappings can induce an output distribution that matches the target distribution. Thus, an adversarial loss alone cannot guarantee that the learned function can map an individual input x_i to a desired output y_i.
This is one of the reasons for introducing the concept of cycle-consistency to produce meaningful mappings, reduce the space of possible mapping functions (can be viewed as a form of regularization). The idea is not to create a pairing between 2 random images which already are in the dataset (the dataset stays unpaired), but to make sure, that if you map a real image from the domain X to the domain Y and then back again, you get the original image back.
Cycle consistency encourages generators to avoid unnecessary changes and thus to generate images that share structural similarity with inputs, it also prevents generators from excessive hallucinations and mode collapse.
I hope that answers your questions.
I'm trying to develop a model to recognize new gestures with the Myo Armband. (It's an armband that possesses 8 electrical sensors and can recognize 5 hand gestures). I'd like to record the sensors' raw data for a new gesture and feed it to a model so it can recognize it.
I'm new to machine/deep learning and I'm using CNTK. I'm wondering what would be the best way to do it.
I'm struggling to understand how to create the trainer. The input data looks like something like that I'm thinking about using 20 sets of these 8 values (they're between -127 and 127). So one label is the output of 20 sets of values.
I don't really know how to do that, I've seen tutorials where images are linked with their label but it's not the same idea. And even after the training is done, how can I avoid the model to recognize this one gesture whatever I do since it's the only one it's been trained for.
An easy way to get you started would be to create 161 columns (8 columns for each of the 20 time steps + the designated label). You would rearrange the columns like
emg1_t01, emg2_t01, emg3_t01, ..., emg8_t20, gesture_id
This will give you the right 2D format to use different algorithms in sklearn as well as a feed forward neural network in CNTK. You would use the first 160 columns to predict the 161th one.
Once you have that working you can model your data to better represent the natural time series order it contains. You would move away from a 2D shape and instead create a 3D array to represent your data.
The first axis shows the number of samples
The second axis shows the number of time steps (20)
The thirst axis shows the number of sensors (8)
With this shape you're all set to use a 1D convolutional model (CNN) in CNTK that traverses the time axis to learn local patterns from one step to the next.
You might also want to look into RNNs which are often used to work with time series data. However, RNNs are sometimes hard to train and a recent paper suggests that CNNs should be the natural starting point to work with sequence data.
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I'm going through this tutorial on RNNs/LSTMs and I'm having quite a hard time understanding stateful LSTMs. My questions are as follows :
1. Training batching size
In the Keras docs on RNNs, I found out that the hidden state of the sample in i-th position within the batch will be fed as input hidden state for the sample in i-th position in the next batch. Does that mean that if we want to pass the hidden state from sample to sample we have to use batches of size 1 and therefore perform online gradient descent? Is there a way to pass the hidden state within a batch of size >1 and perform gradient descent on that batch ?
2. One-Char Mapping Problems
In the tutorial's paragraph 'Stateful LSTM for a One-Char to One-Char Mapping' were given a code that uses batch_size = 1 and stateful = True to learn to predict the next letter of the alphabet given a letter of the alphabet. In the last part of the code (line 53 to the end of the complete code), the model is tested starting with a random letter ('K') and predicts 'B' then given 'B' it predicts 'C', etc. It seems to work well except for 'K'. However, I tried the following tweak to the code (last part too, I kept lines 52 and above):
# demonstrate a random starting point
letter1 = "M"
seed1 = [char_to_int[letter1]]
x = numpy.reshape(seed, (1, len(seed), 1))
x = x / float(len(alphabet))
prediction = model.predict(x, verbose=0)
index = numpy.argmax(prediction)
print(int_to_char[seed1[0]], "->", int_to_char[index])
letter2 = "E"
seed2 = [char_to_int[letter2]]
seed = seed2
print("New start: ", letter1, letter2)
for i in range(0, 5):
x = numpy.reshape(seed, (1, len(seed), 1))
x = x / float(len(alphabet))
prediction = model.predict(x, verbose=0)
index = numpy.argmax(prediction)
print(int_to_char[seed[0]], "->", int_to_char[index])
seed = [index]
model.reset_states()
and these outputs:
M -> B
New start: M E
E -> C
C -> D
D -> E
E -> F
It looks like the LSTM did not learn the alphabet but just the positions of the letters, and that regardless of the first letter we feed in, the LSTM will always predict B since it's the second letter, then C and so on.
Therefore, how does keeping the previous hidden state as initial hidden state for the current hidden state help us with the learning given that during test if we start with the letter 'K' for example, letters A to J will not have been fed in before and the initial hidden state won't be the same as during training ?
3. Training an LSTM on a book for sentence generation
I want to train my LSTM on a whole book to learn how to generate sentences and perhaps learn the authors style too, how can I naturally train my LSTM on that text (input the whole text and let the LSTM figure out the dependencies between the words) instead of having to 'artificially' create batches of sentences from that book myself to train my LSTM on? I believe I should use stateful LSTMs could help but I'm not sure how.
Having a stateful LSTM in Keras means that a Keras variable will be used to store and update the state, and in fact you could check the value of the state vector(s) at any time (that is, until you call reset_states()). A non-stateful model, on the other hand, will use an initial zero state every time it processes a batch, so it is as if you always called reset_states() after train_on_batch, test_on_batch and predict_on_batch. The explanation about the state being reused for the next batch on stateful models is just about that difference with non-stateful; of course the state will always flow within each sequence in the batch and you do not need to have batches of size 1 for that to happen. I see two scenarios where stateful models are useful:
You want to train on split sequences of data because these are very long and it would not be practical to train on their whole length.
On prediction time, you want to retrieve the output for each time point in the sequence, not just at the end (either because you want to feed it back into the network or because your application needs it). I personally do that in the models that I export for later integration (which are "copies" of the training model with batch size of 1).
I agree that the example of an RNN for the alphabet does not really seem very useful in practice; it will only work when you start with the letter A. If you want to learn to reproduce the alphabet starting at any letter, you would need to train the network with that kind of examples (subsequences or rotations of the alphabet). But I think a regular feed-forward network could learn to predict the next letter of the alphabet training on pairs like (A, B), (B, C), etc. I think the example is meant for demonstrative purposes more than anything else.
You may have probably already read it, but the popular post The Unreasonable Effectiveness of Recurrent Neural Networks shows some interesting results along the lines of what you want to do (although it does not really dive into implementation specifics). I don't have personal experience training RNN with textual data, but there is a number of approaches you can research. You can build character-based models (like the ones in the post), where your input and receive one character at a time. A more advanced approach is to do some preprocessing on the texts and transform them into sequences of numbers; Keras includes some text preprocessing functions to do that. Having one single number as feature space is probably not going to work all that well, so you could simply turn each word into a vector with one-hot encoding or, more interestingly, have the network learn the best vector representation for each for, which is what they call en embedding. You can go even further with the preprocessing and look into something like NLTK, specially if you want to remove stop words, punctuation and things like that. Finally, if you have sequences of different sizes (e.g. you are using full texts instead of excerpts of a fixed size, which may or may not be important for you) you will need to be a bit more careful and use masking and/or sample weighting. Depending on the exact problem, you can set up the training accordingly. If you want to learn to generate similar text, the "Y" would be the similar to the "X" (one-hot encoded), only shifted by one (or more) positions (in this case you may need to use return_sequences=True and TimeDistributed layers). If you want to determine the autor, your output could be a softmax Dense layer.
Hope that helps.
What I want to get is: the path which connect all the points in my graph, but without having to tell the algorithm where to start and where to finish.
It need to use the driving direction in google-maps api but without setting a start or end point.
It is not the TSP problem because I don't have a "start city" and I don't have to get back to the "start city" neither.
As expressed in this question: Find the shortest path in a graph which visits certain nodes,
I could just use permutation because I have a few nodes, but the problem is that I need to analyze several groups of this few nodes So I would like the function to be the less time consuming posible.
NOTE: Im not looking for a Minimum Spaning Tree as this one neither: https://math.stackexchange.com/questions/130863/connecting-all-points-on-a-plane-with-shortest-path-possible
I want a path which tell me you will save gas if you go first here, then overthere, then overthere, and finally there.
Question: is there any library which can help me with that? Or is it a know problem that has already an exact answer? How could I solve it?
It sounds like you want an all pairs shortest path algorithm. This is the class of shortest path algorithms that attempt to compute the shortest path (or the length of the shortest path) between every pair of vertices in the graph.
These is a well-known problem, and solutions exist. Here's some reading material that describes other possible algorithms. There might be implementations of Johnson's algorithm for your chosen language and development environment.
Keep in mind, this is an expensive problem, computationally speaking.
If I understand you correctly, you want 1 route to visit all the nodes, without a predefined start/end and you want that to be minimal. A possible solution could be to modify your graph a bit to allow a travelling salesman algorithm to get a complete tour.
You start with your graph and add 1 extra node E. You connect that node to all other nodes in your graph and set the cost of all those edges to a very high constant M. You then unleash a travelling salesman algorithm on that graph which will give you a path P starting at E, passing all nodes and returning to E. If you remove the 2 edges in P that connected E to the rest of your path you will have what you were looking for.
A quick intuitive proof that it is indeed what you were looking for: Suppose it's not the cheapest way to connect all nodes. Let's call the supposedly better path Q. Q and P both connect all nodes in your original graph. The end points of Q would be A and B. Both of these would be connected to node E with an edge of cost M. If you would add those 2 edges to Q, you would get a better TSP solution than P, which is not possible as P was the best.
As you are using google map, your particular instance of TSP might satisfy the Triangle inequality.
Are you really speaking of distances or travel time ?
In the case of distances:
try Googling: "triangle traveling salesman problem"
IMPORTANT: The result is a very good approximation of the best result with guaranteed uper bound, not always the best.
One way to go would be using (self-organized) kohonen networks.
Assume you have n cities on a map (works the same in any dimension).
Take a chain of n connected "neurons" and place it randomly on the map.
Then you do several iterations, one iteration contains:
choose any city. (e.g. go through them in a ordered fashion)
determine the "closest" neuron, call it x. (e.g. euclidian distance)
Move this x closer to the city (e.g. take the direction vector from the neuron to the city and multiply it with a learning rate 0
Move neighbors of this neuron also towards this city (but less than in 3., dependend of distance from the neighbors to the "current closest" neuron x)
One can choose various functions in step 2, 3 and 4.
Notice also that this might not give the globally shortest path since it depends on where the start chain is located and different other things. For this on may consider doing several runs with different starting conditions or (depending of the problem) one can help a bit with pre-knowlege.
I hope this helps to complete this question for further readers...
I need to make an application for creating logic circuits and seeing the results. This is primarily for use in A-Level (UK, 16-18 year olds generally) computing courses.
Ive never made any applications like this, so am not sure on the best design for storing the circuit and evaluating the results (at a resomable speed, say 100Hz on a 1.6Ghz single core computer).
Rather than have the circuit built from the basic gates (and, or, nand, etc) I want to allow these gates to be used to make "chips" which can then be used within other circuits (eg you might want to make a 8bit register chip, or a 16bit adder).
The problem is that the number of gates increases massively with such circuits, such that if the simulation worked on each individual gate it would have 1000's of gates to simulate, so I need to simplify these components that can be placed in a circuit so they can be simulated quickly.
I thought about generating a truth table for each component, then simulation could use a lookup table to find the outputs for a given input. The problem occurred to me though that the size of such tables increase massively with inputs. If a chip had 32 inputs, then the truth table needs 2^32 rows. This uses a massive amount of memory in many cases more than there is to use so isn't practical for non-trivial components, it also wont work with chips that can store their state (eg registers) since they cant be represented as a simply table of inputs and outputs.
I know I could just hardcode things like register chips, however since this is for educational purposes I want it so that people can make their own components as well as view and edit the implementations for standard ones. I considered allowing such components to be created and edited using code (eg dlls or a scripting language), so that an adder for example could be represented as "output = inputA + inputB" however that assumes that the students have done enough programming in the given language to be able to understand and write such plugins to mimic the results of their circuit which is likly to not be the case...
Is there some other way to take a boolean logic circuit and simplify it automatically so that the simulation can determine the outputs of a component quickly?
As for storing the components I was thinking of storing some kind of tree structure, such that each component is evaluated once all components that link to its inputs are evaluated.
eg consider: A.B + C
The simulator would first evaluate the AND gate, and then evaluate the OR gate using the output of the AND gate and C.
However it just occurred to me that in cases where the outputs link back round to the inputs, will cause a deadlock because there inputs will never all be evaluated...How can I overcome this, since the program can only evaluate one gate at a time?
Have you looked at Richard Bowles's simulator?
You're not the first person to want to build their own circuit simulator ;-).
My suggestion is to settle on a minimal set of primitives. When I began mine (which I plan to resume one of these days...) I had two primitives:
Source: zero inputs, one output that's always 1.
Transistor: two inputs A and B, one output that's A and not B.
Obviously I'm misusing the terminology a bit, not to mention neglecting the niceties of electronics. On the second point I recommend abstracting to wires that carry 1s and 0s like I did. I had a lot of fun drawing diagrams of gates and adders from these. When you can assemble them into circuits and draw a box round the set (with inputs and outputs) you can start building bigger things like multipliers.
If you want anything with loops you need to incorporate some kind of delay -- so each component needs to store the state of its outputs. On every cycle you update all the new states from the current states of the upstream components.
Edit Regarding your concerns on scalability, how about defaulting to the first principles method of simulating each component in terms of its state and upstream neighbours, but provide ways of optimising subcircuits:
If you have a subcircuit S with inputs A[m] with m < 8 (say, giving a maximum of 256 rows) and outputs B[n] and no loops, generate the truth table for S and use that. This could be done automatically for identified subcircuits (and reused if the subcircuit appears more than once) or by choice.
If you have a subcircuit with loops, you may still be able to generate a truth table. There are fixed-point finding methods which can help here.
If your subcircuit has delays (and they are significant to the enclosing circuit) the truth table can incorporate state columns. E.g. if the subcircuit has input A, inner state B, and output C, where C <- A and B, B <- A, the truth table could be:
A B | B C
0 0 | 0 0
0 1 | 0 0
1 0 | 1 0
1 1 | 1 1
If you have a subcircuit that the user asserts implements a particular known pattern such as "adder", provide an option for using a hard-coded implementation for updating that subcircuit instead of by simulating its inner parts.
When I made a circuit emulator (sadly, also incomplete and also unreleased), here's how I handled loops:
Each circuit element stores its boolean value
When an element "E0" changes its value, it notifies (via the observer pattern) all who depend on it
Each observing element evaluates its new value and does likewise
When the E0 change occurs, a level-1 list is kept of all elements affected. If an element already appears on this list, it gets remembered in a new level-2 list but doesn't continue to notify its observers. When the sequence which E0 began has stopped notifying new elements, the next queue level is handled. Ie: the sequence is followed and completed for the first element added to level-2, then the next added to level-2, etc. until all of level-x is complete, then you move to level-(x+1)
This is in no way complete. If you ever have multiple oscillators doing infinite loops, then no matter what order you take them in, one could prevent the other from ever getting its turn. My next goal was to alleviate this by limiting steps with clock-based sync'ing instead of cascading combinatorials, but I never got this far in my project.
You might want to take a look at the From Nand To Tetris in 12 steps course software. There is a video talking about it on youtube.
The course page is at: http://www1.idc.ac.il/tecs/
If you can disallow loops (outputs linking back to inputs), then you can significantly simplify the problem. In that case, for every input there will be exactly one definite output. Cycles however can make the output undecideable (or rather, constantly changing).
Evaluating a circuit without loops should be easy - just use the BFS algorithm with "junctions" (connections between logic gates) as the items in the list. Start off with all the inputs to all the gates in an "undefined" state. As soon as a gate has all inputs "defined" (either 1 or 0), calculate its output and add its output junctions to the BFS list. This way you only have to evaluate each gate and each junction once.
If there are loops, the same algorithm can be used, but the circuit can be built in such a way that it never comes to a "rest" and some junctions are always changing between 1 and 0.
OOps, actually, this algorithm can't be used in this case because the looped gates (and gates depending on them) would forever stay as "undefined".
You could introduce them to the concept of Karnaugh maps, which would help them simplify truth values for themselves.
You could hard code all the common ones. Then allow them to build their own out of the hard coded ones (which would include low level gates), which would be evaluated by evaluating each sub-component. Finally, if one of their "chips" has less than X inputs/outputs, you could "optimize" it into a lookup table. Maybe detect how common it is and only do this for the most used Y chips? This way you have a good speed/space tradeoff.
You could always JIT compile the circuits...
As I haven't really thought about it, I'm not really sure what approach I'd take.. but it would possibly be a hybrid method and I'd definitely hard code popular "chips" in too.
When I was playing around making a "digital circuit" simulation environment, I had each defined circuit (a basic gate, a mux, a demux and a couple of other primitives) associated with a transfer function (that is, a function that computes all outputs, based on the present inputs), an "agenda" structure (basically a linked list of "when to activate a specific transfer function), virtual wires and a global clock.
I arbitrarily set the wires to hard-modify the inputs whenever the output changed and the act of changing an input on any circuit to schedule a transfer function to be called after the gate delay. With this at hand, I could accommodate both clocked and unclocked circuit elements (a clocked element is set to have its transfer function run at "next clock transition, plus gate delay", any unclocked element just depends on the gate delay).
Never really got around to build a GUI for it, so I've never released the code.