What kind of equation must I create before arriving at the bit rate calculation if I must perform MFCC math calculations by hand?
Many thanks
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I have data from an accelerometer and made a graph of acceleration(y-axis) and time (x-axis). The frequency rate of the sensor is arround 100 samples per second. but there is no equally spaced time (for example it goes from 10.046,10.047,10.163 etc) the pace is not const. And there is no function of the signal i get. I need to find the frequency of the signal and made a graph of frequency(Hz x-axis) and acceleration (y-axis). but i don't know which code of FFT suits my case. (sorry for bad english)
Any help would be greatly appreciated
For an FFT to work you will need to reconstruct the signal you have with with a regular interval. There are two ways you can do this:
Interpolate the data you already have to make an accurate guess at where the signal would be at a regular interval. However, this FFT may contain significant inaccuracies.
OR
Adjust the device reading from the accelerometer incorporate an accurate timer such that results are always transmitted at regular intervals. This is what I would recommend.
I'm working on a project and I've hit a snag that is past my understanding. My goal is to create an artificial neural network which is fed information from a sound file which is then ported through the system, resulting in a labeling of the chord. I'm hoping to make this to help in music transcription -- not to actually do the transcription itself, but to help in the harmonization aspect. I digress.
I've read as much as I can on the Goertzel and the FFT function, but I'm unsure if these functions are what I'm looking for. I'm not looking for any particular frequency in the sound sample, but rather, I'm hoping to find the higher, middle, and low range frequencies of the sample.
I know the Goertzel algorithm returns a high number if a particular frequency is found, but it seems computational wasteful to run the algorithm for all possible tones in a given sample. Any ideas on what to use?
Or, if this is impossible, I'd love to know that too before spending too much time on this one project.
Thank you for your time!
Probably better suited to DSP StackExchange.
Suppose you FFT a single 110Hz tone to get a spectrogram; you'll see evenly spaced peaks at 110 220 330 etc Hz -- the harmonics. 110 is the fundamental.
Suppose you have 3 tones. Already it's going to look quite messy in the frequency domain. Especially if you have a chord containing e.g. A110 and A220.
On account of this, I think a neural network is a good approach.
Feed in FFT output.
It would be a good idea to use a neural network that accepts complex valued inputs, as FFT outputs of a complex number for each frequency bin.
http://www.eagle.tamut.edu/faculty/igor/PRESENTATIONS/IJCNN-0813_Tutorial.pdf
It may seem computationally wasteful to extract so many frequencies with FFT, but FFT algorithms are extremely efficient nowadays. You should probably use a bit strength of 10, so 2^10 inputs -> 2^9 = 512 complex bins.
FFT is the right solution. Basically, when you have the FFT of an input signal that consists only of sinus waves, you can determine the chord by just mapping which frequencys are present to specific tones in whichever musical temperament you want to use, then look up the chord specified by those tones. If you don't have sinus-waves as input, then using a neural network is a valid attempt in solving the problem, provided that you have enough samples to train it.
FFT is the right way. Harmonics don't bother you, since they are an integer multiple of the fundamental frequency they're just higher 'octaves' of the same note. And to recognize a chord, tranpositions of notes over whole octaves don't matter.
I am trying to estimate the Estimating Quasi-stationary part of a signal in Matlab. It is a 1 second long sound signal that belongs to a bird.
I am using MFCC to extract features but would like to have a window size for MFCC that is guaranteed to operate on statistically quasi-stationary part.
My questions are:
Do you think it is a solid approach if I iterate by varying my window size from 1 second to a smaller interval by observing the change of second moment of features and making a decision where the second moment is not changing anymore?
If I use Shannon entropy method by again varying my MFCC window size, how the number of bits I got at the output of the entropy algorithm would help me to identify the Estimating Quasi-stationary part of the signal
Are there any other ideas?
I have problems when it comes to understand the power spectral density (especially the difference between discrete and continuous). This is probably a "beginner" question but I hope that someone has the time to clarify this for me since literature gave me contradictory results and confused me even more.
I found two terms which are seemingly not the same:
Power density spectrum
Power spectral density (some books say its the scaled power density spectrum)
So in my understanding the power spectral density (Sxx) should give the power per Hz. For a continuous signal it can be calculated with the fourier transform of the autocorrelation function (through the integral the unit would be W/Hz so this seems fine). Followingly the signals power can be calculated by the total integral of Sxx.
But how to calculate it for a discrete and finite signal? ("by hand" and not just use pwelch)
If I apply the autocorrelation and use the fft in this case I obtain: Sxx_disc=1/N*abs(fft(x)).^2 with N=length(x)
But if I use sum(Sxx_disc) it doesn't give me the right signalpower which should be: Px_disc=1/N*sum(abs(x).^2).
This Px_disc I just obtain with sum(Sxx_disc)/N - but why do I have to divide by N again?
Thank you for any kind of help! Klaus
Ok here is what i need to do:
I want to do some tracking using Kalman filter(possibly adaptive).My measurements(when they are available) are very good with very small error from the real measurements. In some cases though the measurements jump to a value,completely off from the correct position i am looking for, and then after few frames the come back to their correct position.
The problem is that if my filter(not adaptive) has specific values for Measurement Noise Covariance(R) and State Error Covariance(Q) matrices the results are not very accurate,because even for these 1% of cases i have to do a compromise between R and Q.
So i decided to use an adaptive Kalman filter as they do in here: http://citeseerx.ist.psu.edu/viewdoc/download?doi=10.1.1.367.1747&rep=rep1&type=pdf
They estimate the measurement noise covariance matrix based on the innovation sequences.
Basically, they are using a moving window on previous samples and the calculate the covariance of the error between the previous measurements-prior estimations. For eg 5 past measurements and the 5 prior estimations.When a faulty measurement comes under the window, the covariance increases and thus the R increases also.
But in practice the R increases(but not enough) so in the next step the estimation is still good but just a bit towards the the faulty measurement.In the next step(because now the the previous estimation has moved a bit towards the measurement) the R becomes smaller with result the new estimation to go even closer to the measurements, and so on and so forth.
In the end after a few frames the estimations follow the faulty measurements. Here is a plot to understand better what i mean.
https://www.dropbox.com/s/rkv0tjcm4s54kv3/untitled.tif
Maybe what i am trying to do is completely wrong and can't be done with the adaptive Kalman filter.Maybe someone who has worked extensively with Kalman Filter in the past and he has faced this problem before can help.
Any idea is welcome!
Before the answer, I want to be sure I got the problem you have right.
You have measurements, some of them are good (Low measurement noise) yet others are outliers.
The problem you're having is tuning the measurement noise covariance matrix.
Practically, you tune for the good measurements.
Outliers measurements are rejected by using the Error Covariance.
If the innovation falls outside an ellipse you define using the Error Covariance Matrix the measurement is rejected.
Whenever a measurement is rejected you just apply the prediction step again and wait for another measurement.
Yes the problem is exactly this.
However i manage to solve it without the need to define any ellipse.What i was doing was correct except the fact that was not working if i had a lot of(lets say fifty) consecutive outliers.
This is normal if you think the size of your window.If it is for example only 10 samples and you have 20 outliers obviously it won't work.But for 5 consecutive outliers work perfectly.Generally i haven't used any threshold as you propose("if the innovation falls outside an ellipse") reject the measurements.I keep the measurements but in the same time when i start to have outliers the Error measurement covariance becomes very large.So the estimation is based more in previous estimation than in current measurement.
If i used your method which is indeed more logical(reject the current measurement,if it is an outlier based on a threshold) i have the problem that i have to define this threshold a priori,right?Maybe i am missing something..