I am currently operating in vhf band AND trying to detect frequencies using Fast Fourier transform thresholding method.
While detection of multiple frequencies , i received spurs(May not appropriate word) in addition with original frequencies, Such as
in case of f1,f2 that are incoming frequencies i also receive their sum f1+f2 and difference f1-f2.
i am trying to eliminate these using thresholding method but i can't differentiate them with real frequency magnitudes.
Please suggest me some method, or methodology to eliminate this problem
Input frequencies F1, F2
Expected frequencies F1,F2
Receive frequencies ,F1,F2,F1-F2,F1+F2
https://imgur.com/3rYYNv2 plot link that elaborate problem
Windowing can reduce windowing artifacts and distant side lobes, but makes the main lobe wider in exchange. But a large reduction in both the main-lobe and near side-lobes normally requires using more data and a longer FFT.
Related
I have a number of time series data sets, which I want to transform to dft signals in order to reduce dimensionality. After transforming to dft, I want to cluster the resulting dft data sets using k-means algorithm.
Since dft signals contain an imaginary number how can one cluster them?
You could simply treat the imaginary part as another component in your vectors. In other applications, you will want to ignore it!
But you'll be facing other, more severe challenges.
Data mining, and clustering in particular, rarely is as easy as appliyng function a (dft) and function b (k-means) and then you have the result, hooray. Sorry - that is not how exploratory data mining works.
First of all, for many time series, DFT will not be helpful at all. On others, you will first have to do appropriate resampling, or segmentation, or get rid of uninteresting effects such as seasonality. Even if DFT works, it may emphasize artifacts such as the sampling frequency or some interferences.
And then you'll run into one major problem: k-means is based on the assumption that all attributes have the same importance. And DFT is based on the very opposite idea: the first components capture most of the signal, the later ones only minor deviations from it (and that is the very motivation for using this as dimensionality reduction).
So based on this intuition, you maybe never should apply k-means on DFT coefficients at all. At the same time, data-mining repeatedly has shown that appfoaches that are "statistical nonsense" can nevertheless provide useful results... so you can try, but verify your resultd with care, and avoid being too enthusiastic or optimistic.
With the help of FFT, it converts dataset into dft signals. It helps to calculates DFT for each small data set.
I am developing a back-end speech recognition software wherein the user can import mp3 files. How can I extract the features from this digital audio file? should I convert it back to analog first?
Your question is unclear, since you are using terms analog and digital incorrectly. Analog is a real-world, continuous function, i.e. voltage, pressure, etc. Digital is a discrete (sampled) and quantized version of the analog signal. You must calculate the FFT of your audio frames when calculating the MFCC's. You can extract MFCC's only from the digital signal - it's rather impossible to do it with the analog one.
If you are asking about whether it is possible to extract the MFCC's from an mp3 file, then yes - it is possible. All you need is to perform the standard algorithm and you can get your features - obviously it is outside of spec of that question.
Calculate the FFT for frames of data.
Calculate the PSD by squaring the samples.
Apply the mel-filterbank and sum the energy across banks.
Calculate the logarithm of each of the energies.
Calculate the DCT of the logarithms of energies.
You're confusing things here, like #jojek said you can do all that WITH the digital signal. This here is a pretty spot on tutorial:
http://practicalcryptography.com/miscellaneous/machine-learning/guide-mel-frequency-cepstral-coefficients-mfccs/
This one is more practical:
http://www.speech.cs.cmu.edu/15-492/slides/03_mfcc.pdf
From Wikipedia: [http://en.wikipedia.org/wiki/Mel-frequency_cepstrum]
MFCCs are commonly derived as follows:[1][2]
Take the Fourier transform of (a windowed excerpt of) a signal. Means short time fourier transform)
Map the powers of the spectrum obtained above onto the mel scale, using triangular overlapping windows. (Calculation described in the links above)
Take the logs of the powers at each of the mel frequencies.
Take the discrete cosine transform of the list of mel log powers, as if it were a signal.
The MFCCs are the amplitudes of the resulting spectrum.
and here's a Matlab toolbox to help you understand it better:
http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html
If I am right, Wavelet packet decomposition (WPT) breaks a signal into various filter banks.
The same thing can be done using many band pass filters.
My aim is to find the energy content of a signal with a large sapmling rate ((2000 hz) in various frequency bands like 1-200, 200-400, 400-600.
What are the advantages and disadvantages of using a WPT of band pass filters?
with wpt (or dwt indeed) you have quadrature mirror filters that will ensure that if you add up all the reconstructed signals in the last level (the leaves) of the wpt tree you get exactly the original signal except for the math processor finite word length aproximations. The algorithm is pretty fast.
Moreover if your signal is non-stationary you can gain the time-frequency localization although this will drastically decrease as you go down on the tree (inverted).
The other aspect is that if yoy are lucky to get a wavelet that correlates well with the non stationary components of your signal the transform will map this components more efficiently.
For you application firstly see how many levels you have to go down in the wpt tree to go from your sampling frequency to the desired freq intervals, you may not get excately 200-400, 400-600 etc,the downer you go in the tree the more accurate are the feq limits, and you may have to join nodes to get your bands.
I'm working on this embedded project where I have to resonate the transducer by calculating the phase difference between its Voltage and Current waveform and making it zero by changing its frequency. Where I(current) & V(Voltage) are the same frequency signals at any instant but not the fixed frequency signals approx.(47Khz - 52kHz). All I have to do is to calculate phase difference between these two signals. Which method will be most effective.
FFT of Two signals and then phase difference between the specific components
Or cross-correlation of two signals?
Or another if any ? Which method will give me most accurate result ? and with what resolution? Does sampling rate affects phase difference's resolution (minimum phase difference which can be sensed) ?
I'm new to Digital signal processing, in case of any mistake, correct me.
ADDITIONAL DETAILS:-
Noise In my system can be white/Gaussian Noise(Not significant) & Harmonics of Fundamental (Which might be significant one in resonant mismatch case).
Yes 4046 can be a good alternative with switching regulators. I'm working with (NCO/DDS) where I can scale/ reshape sinusoidal on ongoing basis.
Implementation of Analog filter will be very complex as I will require higher order filter with high roll-off rate for harmonic removal , so I'm choosing DSP based filter and its easy to work with MATLAB DSP Processors.
What sampling rate would you suggest for a ~50 KHz (47Khz-52KHz) system for achieving result in FFT or Goertzel with phase resolution of preferably =<0.1 degrees or less and frequency steps will vary from as small as ~1 to 2Hz . to 50 Hz-200Hz.
My frequency is variable 45KHz - 55Khz ... But will be known to my system... Knowing phase error for the last fed frequency is more desirable. After FFT AND DIGITAL FILTERING , IFFT can be performed for more noise free samples which can be used for further processing. So i guess FFT do both the tasks ...
But I'm wondering about the Phase difference accuracy cause thats the crucial part.
The Goertzel algorithm http://www.embedded.com/design/configurable-systems/4024443/The-Goertzel-Algorithm is a fairly efficient tone detection method that resolves the signal into real and imaginary components. I'll assume you can do the numeric to get the phase difference or just polarity, as you require.
Resolution versus time constant is a design tradeoff which this article highlights issues. http://www.mstarlabs.com/dsp/goertzel/goertzel.html
Additional
"What accuracy can be obtained?"
It depends...upon what you are faced with (i.e., signal levels, external noise, etc.), what hardware you have (i.e., adc, processor, etc.), and how you implement your solution (sample rate, numerical precision, etc.). Without the complete picture, I'll be guessing what you could achieve as the Goertzel approach is far from easy.
But I imagine for a high school project with good signal levels and low noise, an easier method of using the phase comparator (2 as it locks at zero degrees) of a 4046 PLL www.nxp.com/documents/data_sheet/HEF4046B.pdf will likely get you down to a few degrees.
One other issue if you have a high Q transducer is generating a high-resolution frequency. There is a method but that's another avenue.
Yet more
"Harmonics of Fundamental (Which might be significant)"... hmm hence the digital filtering;
but if the sampling rate is too low then there might be a problem with aliasing. Also, mismatched anti-aliasing filters are likely to take your whole error budget. A rule of thumb of ten times sampling frequency seems a bit low, and it being higher it will make the filter design easier.
Spatial windowing addresses off-frequency issues along with higher roll-off and attenuation and is described in this article. Sliding Spectrum Analysis by Eric Jacobsen and Richard Lyons in Streamlining Digital Signal Processing http://www.amazon.com/Streamlining-Digital-Signal-Processing-Guidebook/dp/1118278380
In my previous project after detecting either carrier, I then was interested in the timing of the frequency changes in immense noise. With carrier phase generation inconstancies, the phase error was never quiescent to be quantified, so I can't guess better than you what you might get with your project conditions.
Not to detract from chip's answer (I upvoted it!) but some other options are:
Cross correlation. Off the top of my head, I am not sure what the performance difference between that and the Goertzel algorithm will be, but both should be doable on an embedded system.
Ad-hoc methods. For example, I would try something like this: bandpass the signals to eliminate noise, find the peaks and measure the time difference between the peaks. This will probably be more efficient, and, provided you do a reasonable job throwing out outliers and handling wrap-around, should be extremely robust. The bandpass filters will, themselves, alter the phase, so you'll have to make sure you apply exactly the same filter to both signals.
If the input signal-to-noise ratios are not too bad, a computually efficient solution can be built based on zero crossing detection. Also, have a look at http://www.metrology.pg.gda.pl/full/2005/M&MS_2005_427.pdf for a nice comparison of phase difference detection algorithms, including zero-crossing ones.
Computing 1-bin of a DFT (or using the similar complex Goertzel block filter) will work if the signal frequency is accurately known. (Set the DFT bin or the Goertzel to exactly that frequency).
If the frequency isn't exactly known, you could try using an FFT with an FFTshift to interpolate the frequency magnitude peak, and then interpolate the phase at that frequency for each of the two signals. An FFT will also allow you to window the data, which may improve phase estimation accuracy if the frequency isn't exactly bin centered (or exactly the Goertzel filter frequency). Different windows may improve the phase estimation accuracy for frequencies "between bins". A Blackman-Nutall window will be better than a rectangular window, but there may be better window choices.
The phase measurement accuracy will depend on the S/N ratio, the length of time one samples the two (assumed stationary) signals, and possibly the window used.
If you have a Phase Locked Loop (PLL) that tracks each input, then you can subtract the phase coefficients (of the generator components) to determine offset between the phases. This would also be robust against noise.
Using GNU octave, I'm computing a fft over a piece of signal, then eliminating some frequencies, and finally reconstructing the signal. This give me a nice approximation of the signal ; but it doesn't give me a way to extrapolate the data.
Suppose basically that I have plotted three periods and a half of
f: x -> sin(x) + 0.5*sin(3*x) + 1.2*sin(5*x)
and then added a piece of low amplitude, zero-centered random noise. With fft/ifft, I can easily remove most of the noise ; but then how do I extrapolate 3 more periods of my signal data? (other of course that duplicating the signal).
The math way is easy : you have a decomposition of your function as an infinite sum of sines/cosines, and you just need to extract a partial sum and apply it anywhere. But I don't quite get the programmatic way...
Thanks!
The Discrete Fourier Transform relies on the assumption that your time domain data is periodic, so you can just repeat your time domain data ad nauseam - no explicit extrapolation is necessary. Of course this may not give you what you expect if your individual component periods are not exact sub-multiples of the DFT input window duration. This is one reason why we typically apply window functions such as the Hanning Window prior to the transform.