(for the quick reader)
Question: Am I right that the Spectral analysis method to analyze a CHIRP is not so beneficial for parameter estimation/ model identification)
[EDIT]
My system is open-loop, 1 input (steering wheel angle) and 2 outputs (y-acceleration and yaw_Rate). To find vehicle characteristics I want to fit a linear transfer function to my data (Bicycle model).
My current method is the 'Spectral analysis method': using test data to estimate the FRF and therefore the transfer function, because:
For dummy data (2 transfer functions excited by a chirp steering wheel angle) this works very well: accuracy of 99.98% to refit the model. For real test data, a real vehicle. this is nowhere near correct. Even if I average the data over 11 runs. Hence my confusion/question.
[will upload images of the test data tonight for clarification]
Background
I'm working on a project where I have to perform parameter identification of a car.
In simulator based compensatory tracking experiments I would excite the 'system' (read human) with a multi-sine signal and use the instrumental variable method (and function fitting) to perform system identification (Fourier transforming in- and output; and only evaluating the excited frequencies).
However, for a human driver this may be a bit difficult to do in the car. It is easier to provide sine-sweep (or CHIRP).
Unfortunately I think this input signal is not compatible with direct frequency domain analysis, because each frequency is only excited during a specific timeframe and the Foerier transform assumes a harmonic oscilation during the entire sample-time.
I have checked some books (System Identification:A Frequency Domain Approach, System identification : an introduction and ) but can't seem to get a grip on how to use the CHIRP signal for the estimation of the Frequency Response Function (thus also the transfer function).
Short answer (for the quick reader):
It depends on what you want to do. And yes, multisine signals can have favourable properties compared to chirps.
A bit longer answer:
You ask about chirp signals and their suitability for system identification / parameter estimation. Hence I assume you focus on frequency domain identification and hence I do not comment on time domain.
If you read the book "System Identification:A Frequency Domain Approach" by Pintelon/Schoukens (try to get the second edition from 2012), you will find (cf chapter 2) that the authors favour periodic signals over nonperiodic ones (like chirps) (and they do for good reasons, as periodic signals avoid major errors like leakage).
However if your system cannot be excited by periodic signals (for whatever reason), chirp signals may be a great excitation signal. In the aviation world, test pilots are even taught to perform good chirp signals. The processing of your data may be different for chirps (take a look at chapter 7 in the Pintelon/Schoukens book).
In the end there is just one thing that makes a good excitation signal - that is it gives the desired estimation result. If chirps work for your application: Go with them!
Unfortunately I think this input signal is not compatible with direct frequency domain analysis, because each frequency is only excited during a specific timeframe and the Foerier transform assumes a harmonic oscilation during the entire sample-time.
I do not understand what you mean with your this paragraph. Can you describe your problem in more detail?
P.S.: You didn't write much about your system. Is it static or dynamic? Linear / Nonlinear? Open loop or closed loop? SISO/MIMO? Are you limited to frequency domain ID? Can you repeat experiments? Each subject should be kept in mind when you decide about the excitation.
Related
The drake multibody plant class has a lot of functions for calculating quantities that I am interested in recording during a simulation (e.g. momentum, center of mass, etc.). What is the best way of recording data like this? I haven't used drake extensively, but I had a few ideas:
Run the simulation in a loop with a defined time step (i.e. simulator.AdvanceTo(current_time + dt)) and use the multibody plant to calculate the quantities directly.
Seems a bit limited (i.e. can't use a single call to AdvanceTo() to run the simulation) and may require a very small time step to get the resolution I'm looking for.
Record available quantities from the multibody plant output ports (e.g. body spatial velocities, body poses, etc.) using a VectorLogSink block, and solve for the quantities of interest after the conclusion of the simulation by reconstructing a Context from the values and calling the multibody plant calculation functions
Not sure if this is possible; seems like a bit of a roundabout approach; the quantities of interest are not available during the simulation
Create a system block that can connect to a multibody plant to perform these computations at every internal simulation timestep. That block could then connect to a VectorLogSink block to record the data.
Not sure if this is possible or where to start
Can anyone provide some guidance on how to record multibody quantities like this?
Porting this question from this issue for continued discussion
Initial response from #jwnimmer-tri
Your assessment of option (1) is correct. If you already know a reasonable dt step size for your logging that suits your needs, it can work well. If the "interesting" times are not always on a fixed schedule, this way can be difficult.
Relatedly, there is an option (4) you hadn't found yet. The Simulator::set_monitor() can be used to set a simulation callback. You could use that for the ability to log at every simulation step, no matter how large or small the step was.
Option (2) should work as well. Assuming that the only state in your simulation is the multibody plant positions and velocities, you could log those during the simulation, and then be able to set the context state to those values offline later, and make any queries on the plant that you need. If you have extra state (e.g., controller state), this approach becomes more difficult.
Option (3) is also possible, but more complicated to explain.
I am working on a problem for which we aim to solve with deep Q learning. However, the problem is that training just takes too long for each episode, roughly 83 hours. We are envisioning to solve the problem within, say, 100 episode.
So we are gradually learning a matrix (100 * 10), and within each episode, we need to perform 100*10 iterations of certain operations. Basically we select a candidate from a pool of 1000 candidates, put this candidate in the matrix, and compute a reward function by feeding the whole matrix as the input:
The central hurdle is that the reward function computation at each step is costly, roughly 2 minutes, and each time we update one entry in the matrix.
All the elements in the matrix depend on each other in the long term, so the whole procedure seems not suitable for some "distributed" system, if I understood correctly.
Could anyone shed some lights on how we look at the potential optimization opportunities here? Like some extra engineering efforts or so? Any suggestion and comments would be appreciated very much. Thanks.
======================= update of some definitions =================
0. initial stage:
a 100 * 10 matrix, with every element as empty
1. action space:
each step I will select one element from a candidate pool of 1000 elements. Then insert the element into the matrix one by one.
2. environment:
each step I will have an updated matrix to learn.
An oracle function F returns a quantitative value range from 5000 ~ 30000, the higher the better (roughly one computation of F takes 120 seconds).
This function F takes the matrix as the input and perform a very costly computation, and it returns a quantitative value to indicate the quality of the synthesized matrix so far.
This function is essentially used to measure some performance of system, so it do takes a while to compute a reward value at each step.
3. episode:
By saying "we are envisioning to solve it within 100 episodes", that's just an empirical estimation. But it shouldn't be less than 100 episode, at least.
4. constraints
Ideally, like I mentioned, "All the elements in the matrix depend on each other in the long term", and that's why the reward function F computes the reward by taking the whole matrix as the input rather than the latest selected element.
Indeed by appending more and more elements in the matrix, the reward could increase, or it could decrease as well.
5. goal
The synthesized matrix should let the oracle function F returns a value greater than 25000. Whenever it reaches this goal, I will terminate the learning step.
Honestly, there is no effective way to know how to optimize this system without knowing specifics such as which computations are in the reward function or which programming design decisions you have made that we can help with.
You are probably right that the episodes are not suitable for distributed calculation, meaning we cannot parallelize this, as they depend on previous search steps. However, it might be possible to throw more computing power at the reward function evaluation, reducing the total time required to run.
I would encourage you to share more details on the problem, for example by profiling the code to see which component takes up most time, by sharing a code excerpt or, as the standard for doing science gets higher, sharing a reproduceable code base.
Not a solution to your question, just some general thoughts that maybe are relevant:
One of the biggest obstacles to apply Reinforcement Learning in "real world" problems is the astoundingly large amount of data/experience required to achieve acceptable results. For example, OpenAI in Dota 2 game colletected the experience equivalent to 900 years per day. In the original Deep Q-network paper, in order to achieve a performance close to a typicial human, it was required hundres of millions of game frames, depending on the specific game. In other benchmarks where the input are not raw pixels, such as MuJoCo, the situation isn't a lot better. So, if you don't have a simulator that can generate samples (state, action, next state, reward) cheaply, maybe RL is not a good choice. On the other hand, if you have a ground-truth model, maybe other approaches can easily outperform RL, such as Monte Carlo Tree Search (e.g., Deep Learning for Real-Time Atari Game Play Using Offline Monte-Carlo Tree Search Planning or Simple random search provides a competitive approach to reinforcement learning). All these ideas a much more are discussed in this great blog post.
The previous point is specially true for deep RL. The fact of approximatting value functions or policies using a deep neural network with millions of parameters usually implies that you'll need a huge quantity of data, or experience.
And regarding to your specific question:
In the comments, I've asked a few questions about the specific features of your problem. I was trying to figure out if you really need RL to solve the problem, since it's not the easiest technique to apply. On the other hand, if you really need RL, it's not clear if you should use a deep neural network as approximator or you can use a shallow model (e.g., random trees). However, these questions an other potential optimizations require more domain knowledge. Here, it seems you are not able to share the domain of the problem, which could be due a numerous reasons and I perfectly understand.
You have estimated the number of required episodes to solve the problem based on some empirical studies using a smaller version of size 20*10 matrix. Just a caution note: due to the curse of the dimensionality, the complexity of the problem (or the experience needed) could grow exponentially when the state space dimensionalty grows, although maybe it is not your case.
That said, I'm looking forward to see an answer that really helps you to solve your problem.
I'm new to machine learning, and I understand that there are parameters and choices that apply to the model you attach to a certain set of inputs, which can be tuned/optimised, but those inputs obviously tie back to fields you generated by slicing and dicing whatever source data you had in a way that makes sense to you. But what if the way you decided to model and cut up your source data, and therefore training data, isn't optimal? Are there ways or tools that extend the power of machine learning into, not only the model, but the way training data was created in the first place?
Say you're analysing the accelerometer, GPS, heartrate and surrounding topography data of someone moving. You want to try determine where this person is likely to become exhausted and stop, assuming they'll continue moving in a straight line based on their trajectory, and that going up any hill will increase heartrate to some point where they must stop. If they're running or walking modifies these things obviously.
So you cut up your data, and feel free to correct how you'd do this, but it's less relevant to the main question:
Slice up raw accelerometer data along X, Y, Z axis for the past A number of seconds into B number of slices to try and profile it, probably applying a CNN to it, to determine if running or walking
Cut up the recent C seconds of raw GPS data into a sequence of D (Lat, Long) pairs, each pair representing the average of E seconds of raw data
Based on the previous sequence, determine speed and trajectory, and determine the upcoming slope, by slicing the next F distance (or seconds, another option to determine, of G) into H number of slices, profiling each, etc...
You get the idea. How do you effectively determine A through H, some of which would completely change the number and behaviour of model inputs? I want to take out any bias I may have about what's right, and let it determine end-to-end. Are there practical solutions to this? Each time it changes the parameters of data creation, go back, re-generate the training data, feed it into the model, train it, tune it, over and over again until you get the best result.
What you call your bias is actually the greatest strength you have. You can include your knowledge of the system. Machine learning, including glorious deep learning is, to put it bluntly, stupid. Although it can figure out features for you, interpretation of these will be difficult.
Also, especially deep learning, has great capacity to memorise (not learn!) patterns, making it easy to overfit to training data. Making machine learning models that generalise well in real world is tough.
In most successful approaches (check against Master Kagglers) people create features. In your case I'd probably want to calculate magnitude and vector of the force. Depending on the type of scenario, I might transform (Lat, Long) into distance from specific point (say, point of origin / activation, or established every 1 minute) or maybe use different coordinate system.
Since your data in time series, I'd probably use something well suited for time series modelling that you can understand and troubleshoot. CNN and such are typically your last resort in majority of cases.
If you really would like to automate it, check e.g. Auto Keras or ludwig. When it comes to learning which features matter most, I'd recommend going with gradient boosting (GBDT).
I'd recommend reading this article from AirBnB that takes deeper dive into journey of building such systems and feature engineering.
I am confused about how linear regression works in supervised learning. Now I want to generate a evaluation function for a board game using linear regression, so I need both the input data and output data. Input data is my board condition, and I need the corresponding value for this condition, right? But how can I get this expected value? Do I need to write an evaluation function first by myself? But I thought I need to generate an evluation function by using linear regression, so I'm a little confused about this.
It's supervised-learning after all, meaning: you will need input and output.
Now the question is: how to obtain these? And this is not trivial!
Candidates are:
historical-data (e.g. online-play history)
some form or self-play / reinforcement-learning (more complex)
But then a new problem arises: which output is available and what kind of input will you use.
If there would be some a-priori implemented AI, you could just take the scores of this one. But with historical-data for example you only got -1,0,1 (A wins, draw, B wins) which makes learning harder (and this touches the Credit Assignment problem: there might be one play which made someone lose; it's hard to understand which of 30 moves lead to the result of 1). This is also related to the input. Take chess for example and take a random position from some online game: there is the possibility that this position is unique over 10 million games (or at least not happening often) which conflicts with the expected performance of your approach. I assumed here, that the input is the full board-position. This changes for other inputs, e.g. chess-material, where the input is just a histogram of pieces (3 of these, 2 of these). Now there are much less unique inputs and learning will be easier.
Long story short: it's a complex task with a lot of different approaches and most of this is somewhat bound by your exact task! A linear evaluation-function is not super-uncommon in reinforcement-learning approaches. You might want to read some literature on these (this function is a core-component: e.g. table-lookup vs. linear-regression vs. neural-network to approximate the value- or policy-function).
I might add, that your task indicates the self-learning approach to AI, which is very hard and it's a topic which somewhat gained additional (there was success before: see Backgammon AI) popularity in the last years. But all of these approaches are highly complex and a good understanding of RL and the mathematical-basics like Markov-Decision-Processes are important then.
For more classic hand-made evaluation-function based AIs, a lot of people used an additional regressor for tuning / weighting already implemented components. Some overview at chessprogramming wiki. (the chess-material example from above might be a good one: assumption is: more pieces better than less; but it's hard to give them values)
this is my first time posting a question here so if the approach is not so standard i apologize, i understand there are lots of questions out there on this and i have read tons of thesis, questions, aritcles and tutorials yet i seem to have a problem and it's always best to ask. i am creating a speech recognition application, using phoneme level processing(not isolated word) continuous HMMs based on gaussian mixture models, involving baum welch, forward-backward, and viterbi algorithms,
i have implemented a very good feature extraction and pre-processing method (MFCC), feature vectors consist of the mfcc, delta and acceleration coefficients as well and it's working pretty well on it's part however when it comes to HMMs , i seem to either have a 'Major Misunderstading' about how HMMs are supposed to help recognize speech or i am missing a little point here...i have try harded a lot that at this point i can't really tell what's wrong and right.
first off, i recorded around 50 words, each 6 utterances, and run them through a correct compatibility and conversion program that i wrote myself and the extracted the features so that they can be used for baum-welch.
i want you to please tell me where am i making a mistake in this procedure, also i will mention a few doubts i have on it so that you can help me understand this whole subject better.
here are the steps in my application concerning anything related to the training :
steps for initial parameters of HMM model :
1 - assign all observations from each training sample of each model to their corresponding discrete state(or in other words, which feature vector belongs to which alphabetic state).
2 - use k-means to find the initial continuous emission parameters, clustering is done over all observations of each state, here the cluster size is 6 (equal to number of mixtures for probability density function), parameters would be sample means, sample covariances and some mixture weights for each cluster.
3 - create initial state-initial and transition probability matrices for each model and training sample individually(left right structure is used in this case), 0 for previous states and 1 for up to 1 next state for transitions and 1 for initial and 0 for others in state initials.
4 - calculate gaussian mixture model based probability density function for each state -> it's corresponding cluster -> assigned to all the vectors in all the training samples for each model
5 - calculate initial emission parameters using the pdf and mixture weights for clusters.
6 - now calculate the gamma variables using initial paramters(transitions, emissions, initials) in forward-backward and initial PDFs, using the continuous formula for gamma..(gamma = probability of being in a certain state at a certain time for any of the mixtures)
7 - estimate new state initials
8 - estimate new state tranisitons
9 - estimate new sample means
10 - estimate new sample covariances
11 - estimate new pdfs
12 - estimate new emissions using new pdfs
repeat the steps from 6 to 12 using new estimated values on each iteration, use viterbi to get an overlook on how the estimating is going and when the probability is not changing anymore, stop and save.
now my issues :
first i don't know if the entire procedure i have followed is correct or not, or is there a better method to approach this...for all i know is that the convergence is pretty fast, for up to 4-5 iterations and it's already not changing anymore, however considering that if i am right then :
it's not possible for me to sit down and pre assign each feature vector to it's state in the beginning at step 1...and i don't think it's a standard procedure either...again i don't even know if i have to do it necessarily, from all my studies it was the best method i could find to get a rapid convergence.
second, say this whole baum welch has done a great job in re estimating and finding local maximums, what's raising my doubt about my baum welch implementation is that how are they later going to help me recognize speech? i assume the estimated parameters are used in viterbi for finding the optimal state for every spoken utterance...if so then emission parameters are not known cause if you look closely you will see that final emission parameters in my algorithm will be assigning each alphabetic state of each model to all the observed signals in each model, other than that...no emission parameters can be found if the signal is not exactly match to the ones used in re-estimation, and it won't obviously work, any attempt to try and match out the signals and find emissions will make the whole HMM lose it's purpose...
again i might have a wrong idea about almost everything here, i would appreciate if you help me understand what i am doing wrong here...if ANYTHING is wrong, notify me please...thank you.
You're attempting to determine the most likely set of phonemes that would have generated the sounds that you're observing - you're not attempting to work out emission parameters, you're working out the most likely set of inputs that would have produced them.
Also, your input corpus is quite small - it's unsurprising that it would converge so quickly. If you're doing this while involved with a university, see if they have access to one of the larger speech corpuses commonly used to train this kind of algorithm on.