Sliding DFT discrete Fourier transform

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SUMMARY

The Sliding Discrete Fourier Transform (DFT) algorithm allows for efficient computation of the DFT for overlapping segments of a signal. By subtracting the first sample and adding the new sample to the existing DFT, users can update the transform without recalculating it from scratch. This technique is particularly useful in real-time signal analysis, such as in music players that display dynamic spectrograms. The algorithm's simplicity and efficiency make it a preferred method for analyzing time-varying signals in various applications.

PREREQUISITES
  • Understanding of Discrete Fourier Transform (DFT)
  • Familiarity with complex exponential functions
  • Basic knowledge of signal processing concepts
  • Experience with real-time data analysis techniques
NEXT STEPS
  • Study the mathematical derivation of the Sliding DFT algorithm
  • Explore applications of the Sliding DFT in real-time audio processing
  • Learn about the Goertzel algorithm and its comparison with Sliding DFT
  • Investigate software tools for implementing Sliding DFT in Python or MATLAB
USEFUL FOR

Signal processing engineers, audio software developers, and researchers interested in real-time signal analysis and visualization techniques.

hxtasy
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"Sliding DFT" discrete Fourier transform...

I was wondering if any of you had had experience with the sliding DFT algorithm. It is somewhat similar to the Goertzel algorithm.

I am having some trouble understanding the mathematics of the algorithm, and I also cannot seem to identify a useful application of it.

So far I have not found a lot of information online and I am hoping somebody that has encountered this in industry can help me out.


thanks,


-Hxtasy
 
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hxtasy said:
I am having some trouble understanding the mathematics of the algorithm, and I also cannot seem to identify a useful application of it.

It sounds like your problems cancel each other out! :wink:

I assume this is what you're talking about: http://www.comm.utoronto.ca/~dimitris/ece431/slidingdft.pdf

I read the paper, and it's really very simple. All it is saying is that if you have, for example, a 10-point DFT of samples 0-9 of a signal (let's call it x(n)), you can turn it into the 10-point DFT for samples 1-10 very easily. All you have to do is subtract x(0) and add x(10) to every point in the DFT, and then multiply each point k (from 0 to 9) in the DFT by e^(k*2*pi*j/10). That's it. Now you can repeat the process to get the DFT for samples 2-11, and so on.

This is useful if you want to analyze a signal by looking at a bunch of DFT's of little chunks in time. This is a very common way of analyzing signals. In fact, where I work, it's almost the only way anyone ever works with a signal. Looking at a set of DFT's for different points in time gives you a spectrum of the signal that changes with time. I can't think of a case where you wouldn't find that information useful. For example, a music player does something like this when it displays a spectrogram while it is playing.
 
Most likely this can only be answered by an "old timer". I am making measurements on an uA709 op amp (metal can). I would like to calculate the frequency rolloff curves (I can measure them). I assume the compensation is via the miller effect. To do the calculations I would need to know the gain of the transistors and the effective resistance seen at the compensation terminals, not including the values I put there. Anyone know those values?

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