I'm trying to implement an Inertial Navigation System using an Indirect Kalman Filter. I've found many publications and thesis on this topic, but not too much code as example. For my implementation I'm using the Master Thesis available at the following link:
https://fenix.tecnico.ulisboa.pt/downloadFile/395137332405/dissertacao.pdf
As reported at page 47, the measured values from inertial sensors equal the true values plus a series of other terms (bias, scale factors, ...).
For my question, let's consider only bias.
So:
Wmeas = Wtrue + BiasW (Gyro meas)
Ameas = Atrue + BiasA. (Accelerometer meas)
Therefore,
when I propagate the Mechanization equations (equations 3-29, 3-37 and 3-41)
I should use the "true" values, or better:
Wmeas - BiasW
Ameas - BiasA
where BiasW and BiasA are the last available estimation of the bias. Right?
Concerning the update phase of the EKF,
if the measurement equation is
dzV = VelGPS_est - VelGPS_meas
the H matrix should have an identity matrix in corrispondence of the velocity error state variables dx(VEL) and 0 elsewhere. Right?
Said that I'm not sure how I have to propagate the state variable after update phase.
The propagation of the state variable should be (in my opinion):
POSk|k = POSk|k-1 + dx(POS);
VELk|k = VELk|k-1 + dx(VEL);
...
But this didn't work. Therefore I've tried:
POSk|k = POSk|k-1 - dx(POS);
VELk|k = VELk|k-1 - dx(VEL);
that didn't work too... I tried both solutions, even if in my opinion the "+" should be used. But since both don't work (I have some other error elsewhere)
I would ask you if you have any suggestions.
You can see a snippet of code at the following link: http://pastebin.com/aGhKh2ck.
Thanks.
The difficulty you're running into is the difference between the theory and the practice. Taking your code from the snippet instead of the symbolic version in the question:
% Apply corrections
Pned = Pned + dx(1:3);
Vned = Vned + dx(4:6);
In theory when you use the Indirect form you are freely integrating the IMU (that process called the Mechanization in that paper) and occasionally running the IKF to update its correction. In theory the unchecked double integration of the accelerometer produces large (or for cheap MEMS IMUs, enormous) error values in Pned and Vned. That, in turn, causes the IKF to produce correspondingly large values of dx(1:6) as time evolves and the unchecked IMU integration runs farther and farther away from the truth. In theory you then sample your position at any time as Pned +/- dx(1:3) (the sign isn't important -- you can set that up either way). The important part here is that you are not modifying Pned from the IKF because both are running independent from each other and you add them together when you need the answer.
In practice you do not want to take the difference between two enourmous double values because you will lose precision (because many of the bits of the significand were needed to represent the enormous part instead of the precision you want). You have grasped that in practice you want to recursively update Pned on each update. However, when you diverge from the theory this way, you have to take the corresponding (and somewhat unobvious) step of zeroing out your correction value from the IKF state vector. In other words, after you do Pned = Pned + dx(1:3) you have "used" the correction, and you need to balance the equation with dx(1:3) = dx(1:3) - dx(1:3) (simplified: dx(1:3) = 0) so that you don't inadvertently integrate the correction over time.
Why does this work? Why doesn't it mess up the rest of the filter? As it turns out, the KF process covariance P does not actually depend on the state x. It depends on the update function and the process noise Q and so on. So the filter doesn't care what the data is. (Now that's a simplification, because often Q and R include rotation terms, and R might vary based on other state variables, etc, but in those cases you are actually using state from outside the filter (the cumulative position and orientation) not the raw correction values, which have no meaning by themselves).
Related
Learner might be in training stage, where it update Q-table for bunch of epoch.
In this stage, Q-table would be updated with gamma(discount rate), learning rate(alpha), and action would be chosen by random action rate.
After some epoch, when reward is getting stable, let me call this "training is done". Then do I have to ignore these parameters(gamma, learning rate, etc) after that?
I mean, in training stage, I got an action from Q-table like this:
if rand_float < rar:
action = rand.randint(0, num_actions - 1)
else:
action = np.argmax(Q[s_prime_as_index])
But after training stage, Do I have to remove rar, which means I have to get an action from Q-table like this?
action = np.argmax(self.Q[s_prime])
Once the value function has converged (values stop changing), you no longer need to run Q-value updates. This means gamma and alpha are no longer relevant, because they only effect updates.
The epsilon parameter is part of the exploration policy (e-greedy) and helps ensure that the agent visits all states infinitely many times in the limit. This is an important factor in ensuring that the agent's value function eventually converges to the correct value. Once we've deemed the value function converged however, there's no need to continue randomly taking actions that our value function doesn't believe to be best; we believe that the value function is optimal, so we extract the optimal policy by greedily choosing what it says is the best action in every state. We can just set epsilon to 0.
Although the answer provided by #Nick Walker is correct, here it's some additional information.
What you are talking about is closely related with the concept technically known as "exploration-exploitation trade-off". From Sutton & Barto book:
The agent has to exploit what it already knows in order to obtain
reward, but it also has to explore in order to make better action
selections in the future. The dilemma is that neither exploration nor
exploitation can be pursued exclusively without failing at the task.
The agent must try a variety of actions and progressively favor those
that appear to be best.
One way to implement the exploration-exploitation trade-off is using epsilon-greedy exploration, that is what you are using in your code sample. So, at the end, once the agent has converged to the optimal policy, the agent must select only those that exploite the current knowledge, i.e., you can forget the rand_float < rar part. Ideally you should decrease the epsilon parameters (rar in your case) with the number of episodes (or steps).
On the other hand, regarding the learning rate, it worths noting that theoretically this parameter should follow the Robbins-Monro conditions:
This means that the learning rate should decrease asymptotically. So, again, once the algorithm has converged you can (or better, you should) safely ignore the learning rate parameter.
In practice, sometimes you can simply maintain a fixed epsilon and alpha parameters until your algorithm converges and then put them as 0 (i.e., ignore them).
I have a series of data and need to detect peak values in the series within a certain number of readings (window size) and excluding a certain level of background "noise." I also need to capture the starting and stopping points of the appreciable curves (ie, when it starts ticking up and then when it stops ticking down).
The data are high precision floats.
Here's a quick sketch that captures the most common scenarios that I'm up against visually:
One method I attempted was to pass a window of size X along the curve going backwards to detect the peaks. It started off working well, but I missed a lot of conditions initially not anticipated. Another method I started to work out was a growing window that would discover the longer duration curves. Yet another approach used a more calculus based approach that watches for some velocity / gradient aspects. None seemed to hit the sweet spot, probably due to my lack of experience in statistical analysis.
Perhaps I need to use some kind of a statistical analysis package to cover my bases vs writing my own algorithm? Or would there be an efficient method for tackling this directly with SQL with some kind of local max techniques? I'm simply not sure how to approach this efficiently. Each method I try it seems that I keep missing various thresholds, detecting too many peak values or not capturing entire events (reporting a peak datapoint too early in the reading process).
Ultimately this is implemented in Ruby and so if you could advise as to the most efficient and correct way to approach this problem with Ruby that would be appreciated, however I'm open to a language agnostic algorithmic approach as well. Or is there a certain library that would address the various issues I'm up against in this scenario of detecting the maximum peaks?
my idea is simple, after get your windows of interest you will need find all the peaks in this window, you can just compare the last value with the next , after this you will have where the peaks occur and you can decide where are the best peak.
I wrote one simple source in matlab to show my idea!
My example are in wave from audio file :-)
waveFile='Chick_eco.wav';
[y, fs, nbits]=wavread(waveFile);
subplot(2,2,1); plot(y); legend('Original signal');
startIndex=15000;
WindowSize=100;
endIndex=startIndex+WindowSize-1;
frame = y(startIndex:endIndex);
nframe=length(frame)
%find the peaks
peaks = zeros(nframe,1);
k=3;
while(k <= nframe - 1)
y1 = frame(k - 1);
y2 = frame(k);
y3 = frame(k + 1);
if (y2 > 0)
if (y2 > y1 && y2 >= y3)
peaks(k)=frame(k);
end
end
k=k+1;
end
peaks2=peaks;
peaks2(peaks2<=0)=nan;
subplot(2,2,2); plot(frame); legend('Get Window Length = 100');
subplot(2,2,3); plot(peaks); legend('Where are the PEAKS');
subplot(2,2,4); plot(frame); legend('Peaks in the Window');
hold on; plot(peaks2, '*');
for j = 1 : nframe
if (peaks(j) > 0)
fprintf('Local=%i\n', j);
fprintf('Value=%i\n', peaks(j));
end
end
%Where the Local Maxima occur
[maxivalue, maxi]=max(peaks)
you can see all the peaks and where it occurs
Local=37
Value=3.266296e-001
Local=51
Value=4.333496e-002
Local=65
Value=5.049438e-001
Local=80
Value=4.286804e-001
Local=84
Value=3.110046e-001
I'll propose a couple of different ideas. One is to use discrete wavelets, the other is to use the geographer's concept of prominence.
Wavelets: Apply some sort of wavelet decomposition to your data. There are multiple choices, with Daubechies wavelets being the most widely used. You want the low frequency peaks. Zero out the high frequency wavelet elements, reconstruct your data, and look for local extrema.
Prominence: Those noisy peaks and valleys are of key interest to geographers. They want to know exactly which of a mountain's multiple little peaks is tallest, the exact location of the lowest point in the valley. Find the local minima and maxima in your data set. You should have a sequence of min/max/min/max/.../min. (You might want to add an arbitrary end points that are lower than your global minimum.) Consider a min/max/min sequence. Classify each of these triples per the difference between the max and the larger of the two minima. Make a reduced sequence that replaces the smallest of these triples with the smaller of the two minima. Iterate until you get down to a single min/max/min triple. In your example, you want the next layer down, the min/max/min/max/min sequence.
Note: I'm going to describe the algorithmic steps as if each pass were distinct. Obviously, in a specific implementation, you can combine steps where it makes sense for your application. For the purposes of my explanation, it makes the text a little more clear.
I'm going to make some assumptions about your problem:
The windows of interest (the signals that you are looking for) cover a fraction of the entire data space (i.e., it's not one long signal).
The windows have significant scope (i.e., they aren't one pixel wide on your picture).
The windows have a minimum peak of interest (i.e., even if the signal exceeds the background noise, the peak must have an additional signal excess of the background).
The windows will never overlap (i.e., each can be examined as a distinct sub-problem out of context of the rest of the signal).
Given those, you can first look through your data stream for a set of windows of interest. You can do this by making a first pass through the data: moving from left to right, look for noise threshold crossing points. If the signal was below the noise floor and exceeds it on the next sample, that's a candidate starting point for a window (vice versa for the candidate end point).
Now make a pass through your candidate windows: compare the scope and contents of each window with the values defined above. To use your picture as an example, the small peaks on the left of the image barely exceed the noise floor and do so for too short a time. However, the window in the center of the screen clearly has a wide time extent and a significant max value. Keep the windows that meet your minimum criteria, discard those that are trivial.
Now to examine your remaining windows in detail (remember, they can be treated individually). The peak is easy to find: pass through the window and keep the local max. With respect to the leading and trailing edges of the signal, you can see n the picture that you have a window that's slightly larger than the actual point at which the signal exceeds the noise floor. In this case, you can use a finite difference approximation to calculate the first derivative of the signal. You know that the leading edge will be somewhat to the left of the window on the chart: look for a point at which the first derivative exceeds a positive noise floor of its own (the slope turns upwards sharply). Do the same for the trailing edge (which will always be to the right of the window).
Result: a set of time windows, the leading and trailing edges of the signals and the peak that occured in that window.
It looks like the definition of a window is the range of x over which y is above the threshold. So use that to determine the size of the window. Within that, locate the largest value, thus finding the peak.
If that fails, then what additional criteria do you have for defining a region of interest? You may need to nail down your implicit assumptions to more than 'that looks like a peak to me'.
These questions regard a set of data with lists of tasks performed in succession and the total time required to complete them. I've been wondering whether it would be possible to determine useful things about the tasks' lengths, either as they are or with some initial guesstimation based on appropriate domain knowledge. I've come to think graph theory would be the way to approach this problem in the abstract, and have a decent basic grasp of the stuff, but I'm unable to know for certain whether I'm on the right track. Furthermore, I think it's a pretty interesting question to crack. So here we go:
Is it possible to determine the weights of edges in a directed weighted graph, given a list of walks in that graph with the lengths (summed weights) of said walks? I recognize the amount and quality of permutations on the routes taken by the walks will dictate the quality of any possible answer, but let's assume all possible walks and their lengths are given. If a definite answer isn't possible, what kind of things can be concluded about the graph? How would you arrive at those conclusions?
What if there were several similar walks with possibly differing lengths given? Can you calculate a decent average (or other illustrative measure) for each edge, given enough permutations on different routes to take? How will discounting some permutations from the available data set affect the calculation's accuracy?
Finally, what if you had a set of initial guesses as to the weights and had to refine those using the walks given? Would that improve upon your guesstimation ability, and how could you apply the extra information?
EDIT: Clarification on the difficulties of a plain linear algebraic approach. Consider the following set of walks:
a = 5
b = 4
b + c = 5
a + b + c = 8
A matrix equation with these values is unsolvable, but we'd still like to estimate the terms. There might be some helpful initial data available, such as in scenario 3, and in any case we can apply knowledge of the real world - such as that the length of a task can't be negative. I'd like to know if you have ideas on how to ensure we get reasonable estimations and that we also know what we don't know - eg. when there's not enough data to tell a from b.
Seems like an application of linear algebra.
You have a set of linear equations which you need to solve. The variables being the lengths of the tasks (or edge weights).
For instance if the tasks lengths were t1, t2, t3 for 3 tasks.
And you are given
t1 + t2 = 2 (task 1 and 2 take 2 hours)
t1 + t2 + t3 = 7 (all 3 tasks take 7 hours)
t2 + t3 = 6 (tasks 2 and 3 take 6 hours)
Solving gives t1 = 1, t2 = 1, t3 = 5.
You can use any linear algebra techniques (for eg: http://en.wikipedia.org/wiki/Gaussian_elimination) to solve these, which will tell you if there is a unique solution, no solution or an infinite number of solutions (no other possibilities are possible).
If you find that the linear equations do not have a solution, you can try adding a very small random number to some of the task weights/coefficients of the matrix and try solving it again. (I believe falls under Perturbation Theory). Matrices are notorious for radically changing behavior with small changes in the values, so this will likely give you an approximate answer reasonably quickly.
Or maybe you can try introducing some 'slack' task in each walk (i.e add more variables) and try to pick the solution to the new equations where the slack tasks satisfy some linear constraints (like 0 < s_i < 0.0001 and minimize sum of s_i), using Linear Programming Techniques.
Assume you have an unlimited number of arbitrary characters to represent each edge. (a,b,c,d etc)
w is a list of all the walks, in the form of 0,a,b,c,d,e etc. (the 0 will be explained later.)
i = 1
if #w[i] ~= 1 then
replace w[2] with the LENGTH of w[i], minus all other values in w.
repeat forever.
Example:
0,a,b,c,d,e 50
0,a,c,b,e 20
0,c,e 10
So:
a is the first. Replace all instances of "a" with 50, -b,-c,-d,-e.
New data:
50, 50
50,-b,-d, 20
0,c,e 10
And, repeat until one value is left, and you finish! Alternatively, the first number can simply be subtracted from the length of each walk.
I'd forget about graphs and treat lists of tasks as vectors - every task represented as a component with value equal to it's cost (time to complete in this case.
In tasks are in different orderes initially, that's where to use domain knowledge to bring them to a cannonical form and assign multipliers if domain knowledge tells you that the ratio of costs will be synstantially influenced by ordering / timing. Timing is implicit initial ordering but you may have to make a function of time just for adjustment factors (say drivingat lunch time vs driving at midnight). Function might be tabular/discrete. In general it's always much easier to evaluate ratios and relative biases (hardnes of doing something). You may need a functional language to do repeated rewrites of your vectors till there's nothing more that romain knowledge and rules can change.
With cannonical vectors consider just presence and absence of task (just 0|1 for this iteratioon) and look for minimal diffs - single task diffs first - that will provide estimates which small number of variables. Keep doing this recursively, be ready to back track and have a heuristing rule for goodness or quality of estimates so far. Keep track of good "rounds" that you backtraced from.
When you reach minimal irreducible state - dan't many any more diffs - all vectors have the same remaining tasks then you can do some basic statistics like variance, mean, median and look for big outliers and ways to improve initial domain knowledge based estimates that lead to cannonical form. If you finsd a lot of them and can infer new rules, take them in and start the whole process from start.
Yes, this can cost a lot :-)
I have to do some work with Q Learning, about a guy that has to move furniture around a house (it's basically that). If the house is small enough, I can just have a matrix that represents actions/rewards, but as the house size grows bigger that will not be enough. So I have to use some kind of generalization function for it, instead. My teacher suggests I use not just one, but several ones, so I could compare them and so. What you guys recommend?
I heard that for this situation people are using Support Vector Machines, also Neural Networks. I'm not really inside the field so I can't tell. I had in the past some experience with Neural Networks, but SVM seem a lot harder subject to grasp. Are there any other methods that I should look for? I know there must be like a zillion of them, but I need something just to start.
Thanks
Just as a refresher of terminology, in Q-learning, you are trying to learn the Q-functions, which depend on the state and action:
Q(S,A) = ????
The standard version of Q-learning as taught in most classes tells you that you for each S and A, you need to learn a separate value in a table and tells you how to perform Bellman updates in order to converge to the optimal values.
Now, lets say that instead of table you use a different function approximator. For example, lets try linear functions. Take your (S,A) pair and think of a bunch of features you can extract from them. One example of a feature is "Am I next to a wall," another is "Will the action place the object next to a wall," etc. Number these features f1(S,A), f2(S,A), ...
Now, try to learn the Q function as a linear function of those features
Q(S,A) = w1 * f1(S,A) + w2*f2(S,A) ... + wN*fN(S,A)
How should you learn the weights w? Well, since this is a homework, I'll let you think about it on your own.
However, as a hint, lets say that you have K possible states and M possible actions in each state. Lets say you define K*M features, each of which is an indicator of whether you are in a particular state and are going to take a particular action. So
Q(S,A) = w11 * (S==1 && A == 1) + w12 * (S == 1 && A == 2) + w21 * (S==2 && A==3) ...
Now, notice that for any state/action pair, only one feature will be 1 and the rest will be 0, so Q(S,A) will be equal to the corresponding w and you are essentially learning a table. So, you can think of the standard, table Q-learning as a special case of learning with these linear functions. So, think of what the normal Q-learning algorithm does, and what you should do.
Hopefully you can find a small basis of features, much fewer than K*M, that will allow you to represent your space well.
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.