I am using fmcw radar to find out distance and speed of moving objects using stm32l476 microcontroller. I transmit the modulation signal as sawtooth waveform and I read the recieved signal in the digital form using ADC function available. Then, I copy this recieved ADC data into fft_in array(converting it into float32_t)(fft_in array size = 512). After copying this fft_in array, I apply fft on this array and process it for finding out range of the object. Until here everything works fine.
Now, in order to find velocity of the object, first, I copy this arrays(fft_in) as rows of the matrix for 64 chirps(Matrix size[64][512]). Then, I take Peak range bin column and apply fft for this column array. So while processing this column array by applying fft, its length reduce to half[32 elements]. Then finding out peak value bin multiplied by frequnecy resolution gives the phase differnce 'w' from which velocity can be calculated as "𝐯=𝛌𝛚/𝟒𝛑𝐓 𝐜".
while running this algorithm, I find that when object is stationery, I get peak value at 22th element(out of 32 elements). what does this imply?
I have sampling frequency for ADC as 24502hz. So per bin value for range estimation is 47.8566hz (24502/512).
I have 64 chirps and Tc is 0.006325s. So 1/0.006325 gives 158.10Hz.What would be per velocity bin resolution, Is it 2.47Hz(158.10/64)? I have bit confusion in this concept.How does 2nd fft works for finding out velocity in fmcw radar?
Infineon has excellent resources on this topic, see this FAQ for the basics: https://www.infineon.com/dgdl/Infineon-Radar%20FAQ-PI-v02_00-EN.pdf?fileId=5546d46266f85d6301671c76d2a00614
If you want to know more details, check out the P2G Software User Manual:
https://www.infineon.com/dgdl/Infineon-P2G_Software_User_Manual-UserManual-v01_01-EN.pdf?fileId=5546d4627762291e017769040a233324 (Chapter 4)
There is even the software available with all the algorithms (including FMCW). How to get the software with the "Infineon Toolbox" is described here: https://www.mouser.com/pdfdocs/Infineon_Position2Go_QS.pdf
Some hints from me:
I suggest applying a window function before the fft https://en.wikipedia.org/wiki/Window_function and remove the mean.
Read about frequency mixers https://en.wikipedia.org/wiki/Frequency_mixer
Related
I've been given some digitized sound recordings and asked to plot the sound pressure level per Hz.
The signal is sampled at 40KHz and the units for the y axis are simply volts.
I've been asked to produce a graph of the SPL as dB/Hz vs Hz.
EDIT: The input units are voltage vs time.
Does this make sense? I though SPL was a time domain measure?
If it does make sense how would I go about producing this graph? Apply the dB formula (20 * log10(x) IIRC) and do an FFT on that or...?
What you're describing is a Power Spectral Density. Matlab, for example, has a pwelch function that does literally what you're asking for. To scale to dBSPL/Hz, simply apply 10*log10([psd]) where psd is the output of pwelch. Let me know if you need help with the function inputs.
If you're working with a different framework, let me know which, 100% sure they'll have a version of this function, possibly with a different output format in which case the scaling might be different.
How to represent a use a sound wave (Sine wave, 1000Hz, 3sec, -3dBFS, 44.1kHz) in FFT program? The input for the program is list of amplitues and sampling rate.
I mean how to transform a sound file(Eg: XYZ.wav file) as input to FFT where one of the input argument needs to take a .dat file consisting of amplitudes and other input arguments needs to take sampling rate and if any necessary.
Typically when you execute a fft call you supply a one dimensional array which represents a curve in the time domain, often this is your audio curve however fft will transform any time series curve ... when you start from an audio file, say a wav file, you must transform the binary data into this floating point 1D array ... if its wav then the file will begin with a 44 byte header which details essential attributes like sample rate, bit depth and endianness ... the rest of the wav file is the payload ... depending on bit depth you will then need to parse a set of bytes then transform the data from typically 16 bit which will consume two bytes into an integer by doing some bit shifting ... to do that you need to be aware of notion of endianness (big or little endian) as well as handling interleaving of a multi-channel signal like stereo ... once you have the generated the floating point array representation just feed it into your fft call ... for starters ignore using a wav file and simply synthesize your own sin curve and feed this into a fft call just to confirm the known frequency in will result in that frequency represented in its frequency domain coming out of the fft call
The response back from a fft call (or DFT) will be a 1D array of complex numbers ... there is a simple formula to calculate the magnitude and phase of each frequency in this fft result set ... be aware of what a Nyquist Limit is and how to fold the freq domain array on top of itself to double the magnitude while only using half of the elements of this freq domain array ... element 0 of this freq domain array is your DC offset and each subsequent element is called a frequency bin which is separated from each other by a constant frequency increment also calculated by a simple formula ... pipe back if interested in what these formulas are
Now you can appreciate people who spend their entire careers pushing the frontiers of working the algos behind the curtain on these api calls ... slop chop slamming together 30 lines of api calls to perform all of above is probably available however its far more noble to write the code to perform all of above yourself by hand as I know it will open up new horizons to enable you to conquer more subtle questions
A super interesting aspect of transforming a curve in time domain into its frequency domain counterpart by making a fft call is that you have retained all of the information of your source signal ... to prove this point I highly suggest you take the next step and perform the symmetrical operation by transforming the output of your fft call back into the time domain
audio curve in time domain --> fft --> freq domain representation --> inverse fft --> back to original audio curve in time domain
this cycle of transformation is powerful as its a nice way to allow you to confirm your audio curve to freq domain is working
First time posting, thanks for the great community!
I am using AudioKit and trying to add frequency weighting filters to the microphone input and so I am trying to understand the values that are coming out of the AudioKit AKFFTTap.
Currently I am trying to just print the FFT buffer converted into dB values
for i in 0..<self.bufferSize {
let db = 20 * log10((self.fft?.fftData[Int(i)])!)
print(db)
}
I was expecting values ranging in the range of about -128 to 0, but I am getting strange values of nearly -200dB and when I blow on the microphone to peg out the readings it only reaches about -60. Am I not approaching this correctly? I was assuming that the values being output from the EZAudioFFT engine would be plain amplitude values and that the normal dB conversion math would work. Anyone have any ideas?
Thanks in advance for any discussion about this issue!
You need to add all of the values from self.fft?.fftData (consider changing negative values to positive before adding) and then change that to decibels
The values in the array correspond to the values of the bins in the FFT. Having a single bin contain a magnitude value close to 1 would mean that a great amount of energy is in that narrow frequency band e.g. a very loud sinusoid (a signal with a single frequency).
Normal sounds, such as the one caused by you blowing on the mic, spread their energy across the entire spectrum, that is, in many bins instead of just one. For this reason, usually the magnitudes get lower as the FFT size increases.
Magnitude of -40dB on a single bin is quite loud. If you try to play a tone, you should see a clear peak in one of the bins.
I have some very short signals from oscilloscope (50k-200k samples) registered over about 2ms time length. Those are acoustic signals with registered signal of a spark of ESD (electrostatic discharge).
I'd like to get some frequency data of that signal, in near-acoustic frequency range (up to about 30kHz) with as high time resolution as possible.
I have tried ploting a spectrogram (specgram in Octave) to view the signal, but the output is not really usefull. Using specgram( x, N, fs );, where x is my signal of fs sampling rate, I receive plot starting at very high frequencies of about 500MHz for low values of N and I get better frequency resolution for big N values (like 2^12-13) but the window is too wide and I receive only 2 spectrum values over whole signal length.
I understand that it may be the limitation of Fourier transform which is probably used by the specgram function (actually, I don't know much about signal analysis).
Is there any other way to get some frequency (as a function of time) information of that kind of signal? I've read something about wavelets, but when I tried using dwt function of signal package, I received this error:
error: 'wfilters' undefined near line 51 column 14
error: called from
dwt at line 51 column 12
Even if this would work, I am not so sure if I'd know how to actually use the output of those wavelet functions ...
To get audio frequency information from such a high sample rate, you will need obtain a sample vector long enough to contain at least a few whole cycles at audio frequencies, e.g. many 10's of milliseconds of contiguous samples, which may or may not be more than your scope can gather. To reasonably process this amount of data, you might low pass filter the sample data to just contain audio frequencies, and then resample it to a lower sample rate, but above twice that filter cut-off frequency. Then you will end up with a much shorter sample vector to feed an FFT for your audio spectrum analysis.
Does anyone know of anywhere I can find actual code examples of Software Phase Locked Loops (SPLLs) ?
I need an SPLL that can track a PSK modulated signal that is somewhere between 1.1 KHz and 1.3 KHz. A Google search brings up plenty of academic papers and patents but nothing usable. Even a trip to the University library that contains a shelf full of books on hardware PLL's there was only a single chapter in one book on SPLLs and that was more theoretical than practical.
Thanks for your time.
Ian
I suppose this is probably too late to help you (what did you end up doing?) but it may help the next guy.
Here's a golfed example of a software phase-locked loop I just wrote in one line of C, which will sing along with you:
main(a,b){for(;;)a+=((b+=16+a/1024)&256?1:-1)*getchar()-a/512,putchar(b);}
I present this tiny golfed version first in order to convince you that software phase-locked loops are actually fairly simple, as software goes, although they can be tricky.
If you feed it 8-bit linear samples on stdin, it will produce 8-bit samples of a sawtooth wave attempting to track one octave higher on stdout. At 8000 samples per second, it tracks frequencies in the neighborhood of 250Hz, just above B below middle C. On Linux you can do this by typing arecord | ./pll | aplay. The low 9 bits of b are the oscillator (what might be a VCO in a hardware implementation), which generates a square wave (the 1 or -1) which gets multiplied by the input waveform (getchar()) to produce the output of the phase detector. That output is then low-pass filtered into a to produce the smoothed phase error signal which is used to adjust the oscillation frequency of b to push a toward 0. The natural frequency of the square wave, when a == 0, is for b to increment by 16 every sample, which increments it by 512 (a full cycle) every 32 samples. 32 samples at 8000 samples per second are 1/250 of a second, which is why the natural frequency is 250Hz.
Then putchar() takes the low 8 bits of b, which make up a sawtooth wave at 500Hz or so, and spews them out as the output audio stream.
There are several things missing from this simple example:
It has no good way to detect lock. If you have silence, noise, or a strong pure 250Hz input tone, a will be roughly zero and b will be oscillating at its default frequency. Depending on your application, you might want to know whether you've found a signal or not! Camenzind's suggestion in chapter 12 of Designing Analog Chips is to feed a second "phase detector" 90° out of phase from the real phase detector; its smoothed output gives you the amplitude of the signal you've theoretically locked onto.
The natural frequency of the oscillator is fixed and does not sweep. The capture range of a PLL, the interval of frequencies within which it will notice an oscillation if it's not currently locked onto one, is pretty narrow; its lock range, over which it will will range in order to follow the signal once it's locked on, is much larger. Because of this, it's common to sweep the PLL's frequency all over the range where you expect to find a signal until you get a lock, and then stop sweeping.
The golfed version above is reduced from a much more readable example of a software phase-locked loop in C that I wrote today, which does do lock detection but does not sweep. It needs about 100 CPU cycles per input sample per PLL on the Atom CPU in my netbook.
I think that if I were in your situation, I would do the following (aside from obvious things like looking for someone who knows more about signal processing than I do, and generating test data). I probably wouldn't filter and downconvert the signal in a front end, since it's at such a low frequency already. Downconverting to a 200Hz-400Hz band hardly seems necessary. I suspect that PSK will bring up some new problems, since if the signal suddenly shifts phase by 90° or more, you lose the phase lock; but I suspect those problems will be easy to resolve, and it's hardly untrodden territory.
This is an interactive design package
for designing digital (i.e. software)
phase locked loops (PLLs). Fill in the
form and press the ``Submit'' button,
and a PLL will be designed for you.
Interactive Digital Phase Locked Loop Design
This will get you started, but you really need to understand the fundamentals of PLL design well enough to build it yourself in order to troubleshoot it later - This is the realm of digital signal processing, and while not black magic it will certainly give you a run for your money during debugging.
-Adam
Have Matlab with Simulink? There are PLL demo files available at Matlab Central here. Matlab's code generation capabilities might get you from there to a PLL written in C.