I recently asked this question:
I am looking for an algorithm to detect pitch. one of the answers suggested that I use an initial FFT to get the basic frequency response, figure out which frequencies are getting voiced, and follow it up with a band pass filter in each area of interest:
A slightly advanced algorithm could do something like this:
Roughly detect pitch frequency (could be done with DFT).
Bandpass signal to filter isolate pitch frequency.
Count the number of samples between two peaks in the filtered signals.
Now I can do the first step okay ( I am coding for iOS, and Apple has a framework (the accelerate framework) for doing FFTs etc.
I have made a start here: but I can see the problem: an FFT that would differentiate all of the possible notes one could sing would require a lot of samples, and I don't want to perform too much unnecessary computation as I'm targeting a mobile device.
So I'm trying to get my head round this answer above, but I don't understand how I could apply the concept of a band pass filter to code.
Can anyone help?
Filter design is pretty complex. There are many techniques. First you have to decide what kind of filter you want to create. Finite impulse response (FIR)? Infinite impulse response (IIR)? Then you select an algorithm for designing a filter of that type. The Remez algorithm is often used for FIR filter design. Go here to see the complexity that I was referring to: http://en.wikipedia.org/wiki/Remez_algorithm
Your best best for creating a filter is to use an existing signal processing library. A quick Google search led me here: http://spuc.sourceforge.net/
Given what your application is, you may want to read about matched filters. I am not sure if they are relevant here, but they might be. http://en.wikipedia.org/wiki/Matched_filter
well in Wikipedia, checkup on low-pass filter, and hi-pass, then join them to make a band-pass filter. Wikipedia has code implementations for those two filters.
http://en.wikipedia.org/wiki/Low-pass_filter
http://en.wikipedia.org/wiki/High-pass_filter
Since you only want to detect a single frequency, it would be an overkill to perform a DFT to then only use one of the values.
You could implement the Goertzel algorithm. Like this C implementation used to detect DTMF tones over a phone line, from the FreePBX source code:
float goertzel(short x[], int nmax, float coeff) {
float s, power;
float sprev, sprev2;
int n;
sprev = 0;
sprev2 = 0;
for(n=0; n<nmax; n++) {
s = x[n] + coeff * sprev - sprev2;
sprev2 = sprev;
sprev = s;
}
power = sprev2*sprev2 + sprev*sprev - coeff*sprev*sprev2;
return power;
}
As you can see, the implementation is fairly trivial and quite effective for single frequencies. Check the link for different versions with and without floating point, and how to use it.
Related
The N2 diagram for my full problem is below.
The N2 diagram for the coupled portion of the problem is below.
I have a DirectSolver handling the coupling between LLTForces and ImplicitLiftingLine, and an LNBGS solver handling the coupling between LiftingLineGroup and TestCL.
The gist for the problem is here: https://gist.github.com/eufren/31c0e569ed703b2aea3e2ef5360610f7
I have implemented guess_nonlinear() on ImplicitLiftingLine, which should use various outputs from LLTGeometry to give a good initial guess for the vortex strengths based on a linearised form of the governing equations.
def guess_nonlinear(self, inputs, outputs, resids):
freestream_unit_vector = inputs['freestream_unit_vector']
freestream_velocity = inputs['freestream_velocity']
n = inputs['normal_vectors']
A = inputs['surface_areas']
l = inputs['bound_vortices']
ic_tot = inputs['influence_coefficients_total']
v_inf = freestream_velocity
v_inf_vec = v_inf*freestream_unit_vector
lin_numerator = np.pi * v_inf * A * np.sum(n * v_inf_vec, axis=1)
lin_denominator = (np.linalg.norm(np.cross(v_inf_vec, l), axis=1) - np.pi * v_inf * A * np.sum(np.sum(n * ic_tot, axis=2), axis=1))
lin_vtx_str = lin_numerator / lin_denominator
outputs['vortex_strengths'] = lin_vtx_str
However, when the problem is run for the first time, any inputs not explicitly set with p.set_val() are all 1s. This causes guess_nonlinear() to give a bad output and so the system fails to converge:
As far as I can tell, the execution order for the LLT group is correct, and the geometry components should be being executed before the implicit component. I'm confused as to why this doesn't seem to actually be happening when the code is run, and instead these inputs are taking their default values.
What do I need to change to get this to work properly? Additionally, I've found difficulty in getting LNBGS to converge (hence adding guess_nonlinear()) during optimisation - only DirectSolver gets all the way through the optimisation without issues, but it's very slow for large numbers of LLT nodes). How can I improve the linear and nonlinear solver selection, and improve the reliability of the iterative solver?
Note: Thanks for providing a testable example. It made figuring out the answer to your question a lot simpler. Your problem was a bit subtle and I would not have been able to give a good answer without runnable code
Your first question: "Why are all the inputs 1"
"Short" Answer
You have put the nonlinear solver to high in the model hierarchy, which then included a key precurser component that computed your input values. By moving the solver down to a lower level of the model, I was able to ensure that the precurser component (LTTGeometry) ran and had valid outputs before you got to the guess_nonlinear of implicit component.
Here is what you had (Notice the implicit solver included LTTGeometry even though the data cycle does not require that component:
I moved both the nonlinear solver and the linear solver down into the LTTCycle group, which then allows the LTTGeometry component to execute before getting to the nonlinear solver and guess_nonlinear step:
My fix is only partially correct, since there is a secondary cycle from the TestCL component that also needs a solver and does not have one. However, that cycle still does not involve the LTTGeometry group. So the fully correct fix is to restructure you model top run geometry first, and then put the LTTCycle and TestCL groups together so you can run a solver over just them. That was a bit more hacking than I wanted to do on your test problem, but you can see the general idea from the adjusted N2 above.
Long Answer
The guess_nonlinear sequence in OpenMDAO does NOT run the compute method of explicit components or of groups. It follows the execution hierarchy, and calls any guess_nonlinear that it finds. So that means that any explicit components you have in your model will NOT get executed, their outputs will not get updated with computed values, and those computed values will not get passed to the inputs of downstream components.
Things get a little tricky when you have deep model hierarchies. The guess_nonlinear method is called as the first step in the nonlinear solver process. If you have a NonLinearRunOnce solver at the top level, it will follow the compute chain down the line calling compute or solve_nonlinear on each child and doing a data transfer after each one. If one of those children happens to be a group with a nonlinear solver, then that solver will call guess_nonlinear on its children (grandchildren of the top group with the NonLinearRunOnce solver) as the first step. So any outputs that were computed by the siblings of this group will be valid, but none of the outputs from the grandchild level will have been computed yet.
You may be wondering why not just have the guess_nonlinear method call the compute for any explicit components? There is a difficult to balance trade off here. If you assume that all explicit components are very cheap to run, then it might make sense to run the compute methods --- or it might not. A lot depends on the cyclic data structure. If some early component in the group needs guesses from the later one, then running its compute isn't going to help you much at all. Perhaps more importantly though, not all explicit components are cheap to run. You might have a very expensive computation, and calling compute as part of the guess process would be way too costly.
The compromise here, if you need some kind of top level guess process, is that you can implement guess_nonlinear at the group level. It's less common to do, but it gives you total control over what happens. You can call whatever you need to call in whatever sequence.
So the absolute key thing to remember is that the only data you have available to you when a guess_nonlinear is called is any data that was computed before your containing solver was executed. That means any thing that was computed before you got to the model scope of the containing solver (not the scope of the component with the guess_method itself).
Your second question: "How can I speed this up when the number of nodes gets large?"
This one not possible to give a generic answer to at all. I noticed that you have already specified sparse partial derivatives. That is a great start, but if its still not fast enough for you then it means you're reaching the limits of what you can do with a DirectSolver. You note that this solver is the only one that gets you through the optimization without issues, which I will take to mean that ScipyKryloventer link description here and PetscKrylov are not converging the linear system well for you --- at least not by themselves. Thats not surprising, as krylov solvers almost always require some kind of preconditioner... and this is why I can't offer a generic answer. Setting up efficient linear solvers for larger-scale compute is a tricky subject. If you look into the literature, you'll find some good suggestions. You can also study open source implementations like VSPAero for some tips.
effectively, you've reached the limit of what simple linear solvers can offer you. From this point forward, OpenMDAO can help a bit by making it easier to implement some preconditioning, but you'll have to suffer the math side yourself.
Using Windows API, I want to implement something like following:
i.e. Getting current microphone input level.
I am not allowed to use external audio libraries, but I can use Windows libraries. So I tried using waveIn functions, but I do not know how to process audio input data in real time.
This is the method I am currently using:
Record for 100 milliseconds
Select highest value from the recorded data buffer
Repeat forever
But I think this is way too hacky, and not a recommended way. How can I do this properly?
Having built a tuning wizard for a very dated, but well known, A/V conferencing applicaiton, what you describe is nearly identical to what I did.
A few considerations:
Enqueue 5 to 10 of those 100ms buffers into the audio device via waveInAddBuffer. IIRC, when the waveIn queue goes empty, weird things happen. Then as the waveInProc callbacks occurs, search for the sample with the highest absolute value in the completed buffer as you describe. Then plot that onto your visualization. Requeue the completed buffers.
It might seem obvious to map the sample value as follows onto your visualization linearly.
For example, to plot a 16-bit sample
// convert sample magnitude from 0..32768 to 0..N
length = (sample * N) / 32768;
DrawLine(length);
But then when you speak into the microphone, that visualization won't seem as "active" or "vibrant".
But a better approach would be to give more strength to those lower energy samples. Easy way to do this is to replot along the μ-law curve (or use a table lookup).
length = (sample * N) / 32768;
length = log(1+length)/log(N);
length = max(length,N)
DrawLine(length);
You can tweak the above approach to whatever looks good.
Instead of computing the values yourself, you can rely on values from Windows. This is actually the values displayed in your screenshot from the Windows Settings.
See the following sample for the IAudioMeterInformation interface:
https://learn.microsoft.com/en-us/windows/win32/coreaudio/peak-meters.
It is made for the playback but you can use it for capture also.
Some remarks, if you open the IAudioMeterInformation for a microphone but no application opened a stream from this microphone, then the level will be 0.
It means that while you want to display your microphone peak meter, you will need to open a microphone stream, like you already did.
Also read the documentation about IAudioMeterInformation it may not be what you need as it is the peak value. It depends on what you want to do with it.
Have to say surprised with the limitations of labVIEW FPGA in array implementation. I am not experienced enough in labview to make this comment but I found it very difficult to work with arrays !!.
I am working on Active Noise Cancellation project. I need to collect audio data from two microphones #40k Sample rate and 100 samples per frame (at least) and output the audio through loud speaker #40k sample rate. For this purpose, I am using myRIO 1900 in FPGA High throughput personality.
Right now, I am trying to implement LMS algorithm in LabVIEW FPGA platform. I have attached the MATLAB code below, what I want to implement !
Till array intializations are fine but when it comes to temporary vector manipulations. like :
x_temp(1:nr_c-1)=x_temp(2:nr_c);
x_temp(nr_c)=x(i);
I am going mad ! How to do this ? LabVIEW doesnt allow us to index the 1D arrays properly ( we can only extract one particular element of array not part of array, I can extract x_temp(1:nr_c) with sub array function but how to extract x_temp(2:nr_c) ??)
Please help me how to do the manipulation of 'x_temp' vector with the above basic statements.
PS: 1.LabVIEW FPGA supports only one dimensional array operations.
Even though Adaptive Filter Toolkit pallettes are available in myRIO FPGA High throughput personality, I cannot install Adaptive Filter Toolkit on myRIO Target.!!!
3.In MyLMS program : x : input vector(Array) , y: Desired Signal, nr_c : number of filter coefficients ,step: step size. I hope you understood the rest program :)
MyLMS.m :
function [y_hat,e,w] = MyLms(step, nr_c, x, y)
%intializing all vectors
coeffs=zeros(1,nr_c);
x_temp=zeros(1,nr_c);
y_hat=zeros(length(x),1);
e=zeros(length(x),1);
for i=1:length(x)
%temporary vector formation
x_temp(1:nr_c-1)=x_temp(2:nr_c);
x_temp(nr_c)=x(i);
%LMS algorithm iterations
y_hat(i) = x_temp * coeffs';
e(i)= y(i) - y_hat(i);
coeffs = coeffs+step*e(i)*x_temp;
end
I am by no means a LabVIEW FGPA expert, but I think you would encounter very similar limitations in VHDL.
At any rate, the typical technique for accessing subarrays is using rotate and replace subset (see Joel_G's post). For more information about the FPGA constraints and capabilities, see the help document: Array Palette Details (FPGA Module).
I have a problem with recognising a signal. Let say the signal is a quasiperiodic signal, the period time has finite limits. The "shape" of the signal must match some criteria, so the actual algorithm using signal processing techniques such as filtering, derivating the the signal, looking for maximum and minimum values. It has a good rate at finding the good signals, but the problem is it also detects wrong shapes too.
So I want to use Aritifical Intelligence - mainly Neural Networks - to overcome this problem. I thought that a multi layer network with some average inputs (the signal can be reduced) and one output whould shows the "matching" from 0..1. However the problem is that I never did such a thing, so I am asking for help, how to achive something like this? How to teach the neural network to get the expected results? (let say I have vectors for inputs which should give 1 as output)
Or this whole idea is a wrong approximation for the problem? I am open to any learning algorithms or idea to learn and use to overcome on this problem.
So here is a figure on the measured signal(s) (values and time is not a concern now) and you can see a lot "wrong" ones, the most detected signals are good as mentioned above.
Your question can be answered in a broad manner. You should consider editing it to prevent it to be closed.
But anyway, Matlab had a lot of built-in function and toolbox to support Artificial Intelligence, with a lot of sample code available, which you can modify and refer to. You can find some in Matlab FileExchange.
And I know reading a lot of technical paper for Artificial Intelligence is a daunting task, so good luck!
You can try to build a neural network using Neuroph. You can inspire from "http://neuroph.sourceforge.net/TimeSeriesPredictionTutorial.html".
On the other hand, it is possible to approximate the signal using Fourier transformation.
You can try 1D convolution. So the basic idea is you give a label 0: bad, 1: good to each signal value at each timestamp. After this, you can model
model = Sequential()
model.add(Conv1D(filters=64, kernel_size=3, activation='relu', padding = 'same', input_shape=(1,1)))
model.add(Bidirectional(LSTM(20, return_sequences=True)))
model.add(Conv1D(filters=64, kernel_size=3, activation='relu', padding = 'same'))
model.add(Dropout(0.5))
model.add(Flatten())
model.add(Dense(100, activation='sigmoid'))
model.add(Dense(2, activation='softmax'))
model.compile(loss='binary_crossentropy', optimizer='adam', metrics=['accuracy'])
Train the model and then give it a new signal to predict. It will predict given series to 0 and 1 values. if count of 0 is more than count of 1, the signal is not good.
am trying to calculate mean and variance using 3X3 window over image(hXw) in opencv...here is my code...is there any accuracy issues with this??or is there any other efficient method to do it in one pass.?
int pi,a,b;
for(i=1;i<h-1;i++)
{
for(j=1;j<w-1;j++)
{ int sq=0,sum=0;
double mean=0;
double var=0;
for(a=-1;a<=1;a++)
{
for(b=-1;b<=1;b++)
{
pi=data[(i+a)*step+(j+b)];
sq=pi*pi;
sum=sum+sq;
mean=mean+pi;
}
}
mean=mean/9;
double soa=mean*mean;//square of average
double aos=sum/9;//mean of squares
double var=aos-soa;//variance
}
}
With respect to computational efficiency I would recommend doing this in the Fourier domain instead of the time (image) domain using convolutions. Remember, a convolution is a simple multiplication in the Fourier domain. Just like in time series where the spectral density function is the variance decomposed as a function of frequency, one can extend this into two dimensions for an image. Should be much better than nested for-loops.
I don't have the code on me at the moment. but this technique has been used in algorithms like "fast template matching" for object detection or image registration.
That is a pretty well-researched topic, see e.g. this Wikipedia article on variance calculations.
One of the issues that sometimes gets mentioned is accumulated numerical errors; you need to decide if that may be an issue. If the values you compute over are similar in range that it may be less of an issue.
You should be fine even with floats over such a small number of pixels. Typically you need doubles if you're doing this kind of thing over an entire image.
You should better use image integrals for quick local mean and standard deviation calculation!
All you need in that case is to correctly calculate the boundaries of the mask window at each position of the image. It will be much more faster.
If you will need a sample code, please ask for that.