When using a FFT should I convert the AtoD samples into a Voltage?

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Discussion Overview

The discussion revolves around the use of Fast Fourier Transform (FFT) in digital signal processing (DSP) and whether A to D (analog to digital) samples should be converted back into voltages before performing the FFT. Participants explore the implications of scaling factors, the handling of negative values, and the treatment of AC signals from microphones.

Discussion Character

  • Exploratory
  • Technical explanation
  • Debate/contested

Main Points Raised

  • One participant questions whether A to D samples should be converted to voltages before applying the FFT, expressing uncertainty about the necessity of this conversion.
  • Another participant states that the units are a scaling factor and do not change the relative values of amplitudes, emphasizing the importance of consistency when comparing spectra.
  • It is suggested that computations should be performed in the digital domain, with scaling to the analog domain only if necessary for specific applications, such as determining energy levels of signals.
  • A participant raises a concern about handling AC signals from microphones, specifically regarding the need to offset the zero crossing voltage for ADC input and the implications for FFT processing.
  • There is mention of 2's complement representation of A/D numbers from sound cards, with a suggestion that this format allows for straightforward processing without conversion to voltages.

Areas of Agreement / Disagreement

Participants express differing views on whether to convert A to D samples to voltages before performing FFT. Some advocate for keeping samples in the digital domain, while others highlight the importance of scaling for consistency in comparisons. The discussion remains unresolved regarding the best approach to handling AC signals and the implications of ADC output formats.

Contextual Notes

Participants note potential issues with normalization factors in FFT processing and the handling of DC offsets in digital data, indicating that these aspects may complicate the analysis.

Who May Find This Useful

This discussion may be useful for individuals involved in digital signal processing, particularly those working with audio signals and FFT analysis, as well as those interested in the implications of A to D conversion techniques.

btb4198
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when using the FFT for DSP should I convert the A to D samples back into a Voltages ?

so

A to D sample = sample voltage * ( bits / max Voltages)

so

sample Voltage = (A to D sample ) *(max Voltages/ bits)
that is right, right?

anyhow should I keep the samples as digital samples or convert them to analog samples
before I run them into the FFT?
why or why not?

I kind of think it should matter... but I do not know . I am new to this.

I do think that the A to D / mic on a computer might remove all the negative numbers, but i do not know is you would get negative numbers from a microphone...
do you ?
 
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The units are only a scaling factor ... they don't change the relative values of the amplitudes.

They would be important if you were comparing one spectrum with another: you need to be consistent.
 
Do all your computations in the digital domain, and afterwards scale your results to the analog domain if you need to. For example you may need to know the actual energy level of a modulated signal within a bandwidth.

I get in trouble with the 1/N normalization thing all the time. Should I use 1/N for forward, inverse, or 1/sqrt(N) on both.
 
btb4198 said:
when using the FFT for DSP should I convert the A to D samples back into a Voltages ?

so

A to D sample = sample voltage * ( bits / max Voltages)

so

sample Voltage = (A to D sample ) *(max Voltages/ bits)
that is right, right?

anyhow should I keep the samples as digital samples or convert them to analog samples
before I run them into the FFT?
why or why not?

I kind of think it should matter... but I do not know . I am new to this.

I do think that the A to D / mic on a computer might remove all the negative numbers, but i do not know is you would get negative numbers from a microphone...
do you ?

A microphone puts out an AC signal. How are you handling that at your ADC? Are you offsetting the zero crossing voltage of the input signal to bias it up to half of the input voltage range of the ADC? If so, you will want to subtract that offset out of your digital data before doing the FFT, or else just subtract the DC component out of the final FFT data.

If your ADC circuit outputs 2's complement data to represent the AC waveform, you will probably need to do something before the FFT, I would think...
 
If you get A/D numbers from a sound card or wave file they are 2's complement positive and negative numbers. Is that what you are doing? Just crunch on them to your hearts content.
 
Last edited:
yes, I am getting them from a sound card on my computer ...
 

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