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signal processing is not my domain of expertise , but I need to make use of it, to do an HRV analysis. These are steps I do starting from my raw ECG signal.

ECG signal -> RR peak detection -> find RR intervals -> cubic spline interpolation -> IIR high pass filter -> FFT

The peaks I get is a mirrored image, I guess its because its a pure signal from a digital patient simulator like all at 60 bpm or all at 120 bpm etc and follows the rule of Hermetian symmetry.

However my FFT output at pure 60 bpm should have indicated a peak at 1 Hz and a 120bpm at 2 Hz, is what I understood. But my all peaks start from 0.

Please help

PS : Implementing this all in an android application.

PS : This is what my output looks on a 60 bpm pure signal.

Three graphs in order : RR interval, RR after interpolation, FFT

PS : Added log data

FFT output : http://www.yourfilelink.com/get.php?fid=1337132

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  • $\begingroup$ so what's your domain of expertise ? $\endgroup$ – Fat32 Mar 23 '17 at 8:17
  • $\begingroup$ Android development $\endgroup$ – Diwesh Saxena Mar 23 '17 at 8:34
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The FFT output will be bins evenly spaced from DC (bin 0) to 1 bin less than your sampling frequency. Thus your sampling frequency is at N for bins 0 to N-1, where N is the number of samples in your time domain sequence.

Within the first Nyquist zone of $-f_s/2$ to $+f_s/2$ where $f_s$ is the sampling rate, the bins from 0 to N/2-1 are the positive frequency components, while the bins from N/2 to N-1 are the negative components. Specifically bin N-1 is -1, bin N-2 is -2, etc so you could view the domain as a circle from 0 to N-1 and repeating again.

The reason your peaks are a mirror image is because your time domain signal is real, and for real signals the positive and negative frequencies have a conjugate symmetric relationship (same magnitude, opposite phase).

The reason you see a large response in bin 0 is because your time domain signal has a non-zero mean. Bin 0, being the DC term, is the mean of your signal (scaled by a factor of N). You likely don't care about DC, and would therefore benefit by removing the mean of your signal before taking the FFT.

The FFT is an algorithm that implements the DFT efficiently, which is as the name implies a discrete version of the Fourier Transform. I think it is helpful to see that the Fourier Transform is simply a correlation as we sweep frequency. (Where correlation is complex conjugate multiply and integrate, or equivalently discrete as complex conjugate multiply and accumulate). Thus for the DFT, each bin is a correlation of a single rotating phasor (your traditional sine wave is made of two equal and opposite rotating phasors as shown by Euler's Identity: $\frac{1}{T}(e^{j\omega t}+e^{-j\omega t}$). So for the DFT, given as: $$X[k]=\sum_{n=0}^{N-1}x[n]e^{-j(k\omega_0) n}$$

So we see that for the first bin (k=0), we just have N times the mean of x. If x has a DC value, we would get a strong correlation for that calculation. Also observe when k =1, which corresponds to the next bin, the exponential is spinnning at its lowest possible rate $\omega_0$ corresponding to one cycle over your time domain sequence. So in that calculation we are computing the correlation of the lowest possible frequency that is spinning in the opposite direction ($e^{+j\omega t}$): such a signal in our x(t), after this computation would have the same result as bin 0: we would get N times the mean after having "derotated" the signal that was present there. Thus every bin goes through the same "averaging filter" which is a Sinc function in frequency (not a great filter), so our output at any given bin will be sensitive to some degree of frequencies everywhere in our spectrum (and as we see with the Sinc function there will be discrete frequencies that are at null positions and therefore will be locations where there is no "leakage" into other bins).

This should give you enough insight into some important fundamental points to dig further into the DFT/FFT. In addition here are some other related posts I think will help you:

Frequency measurement from acoustic emission sensor data(VDC RMS)

What proportion of a padded FFT should be actual values

Signal Processing/FFT gives very high magnitudes for low frequencies

You will want to read up a little more about windowing too; the links above and other posts (by search) include more information on that. Ultimately fred harris' paper has been a great resource for me on that topic.

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  • $\begingroup$ Thanks for the useful resources and great explanation. So when you say I should essentially remove the DC components before applying the FFT, are you indirectly saying applying a high pass filter before FFT ? Does it do the same thing ? Also I added my output please see $\endgroup$ – Diwesh Saxena Mar 24 '17 at 0:04
  • $\begingroup$ Not really, what I suggest is actually a bad way to do a high pass filter but is very simple: the first bin is this mean value so we are essentially removing that; but generally not the way I would recommend filtering. (There are other posts that detail that but is essentially the "frequency sampling" method of filter design $\endgroup$ – Dan Boschen Mar 24 '17 at 0:10
  • $\begingroup$ Hmm, well since my knowledge is limited, I did try to apply an IIR second order high pass filter and then passed the output of that to FFT. As you can see from my graph the result of a 60 bpm data attached, doesnt seem right.Looking at the image can you think of probably where did I go wrong ? $\endgroup$ – Diwesh Saxena Mar 24 '17 at 0:26
  • $\begingroup$ Try my method if you didn't yet as although not ideal will be a little more fool proof $\endgroup$ – Dan Boschen Mar 24 '17 at 0:49
  • $\begingroup$ I did , removing the first bin and then plotting. bt as you can see from the graph I attached, even if i remove the first half, the graph still plots from max peak at 0, which is why I am so confused :( $\endgroup$ – Diwesh Saxena Mar 24 '17 at 0:54

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