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Recover the original analog signal (time varying Voltage) from digitized version?

开发者 https://www.devze.com 2023-02-27 01:44 出处:网络
I have been looking into how to convert my digital data into analog. So, I have a two column ASCII data file (x: time, y=voltage amplitude) which I would like to convert into an analog signal (varyi

I have been looking into how to convert my digital data into analog.

So, I have a two column ASCII data file (x: time, y=voltage amplitude) which I would like to convert into an analog signal (varying Voltage with time). There are Digital to Analog converters, but the good ones are quite expensive. There should be a more tr开发者_StackOverflow中文版ivial way to achieve this.

Ultimately what I'd like to do is to reconstruct the original time variant voltage which was sampled every nano-second and recorded as an ASCII data file.

I thought I may feed the data into my laptop's sound card and re-generate the time variant voltage which I can then feed into the analyzer via the audio jack. Does this sound feasible?

I am not looking into recovering the "shape" but the signal (voltage) itself.


Puzzled on several accounts.

You want to convert into an analog signal (varying Voltage with time) But the what you already have, the discrete signal, is indeed a "varying voltage with time", only that both the values (voltages) and times are discrete. That's the way computers (digital equipment, in general) work.

Only when the signal goes to some non-discrete medium (eg. a classical audio cable+plug) we have an analog signal. Precisely, the sound card of your computer is at its core a "Digital to Analog converter".

So, it appears you are not trying to do some digital processing of your signal (interpolation, or whatever), you are not dealing with computer programming, but with a hardware thing: getting the signal to a cable. If so, SO is not the proper place. YOu might try https://electronics.stackexchange.com/ ...

But, on another thing, you say that your data was "sampled every nano-second". That means 1 billion samples per second, or a sample freq of 1Ghz. That's a ridiculously high frequency, at least in the audio world. You cant output that to a sound card, which would be limited to the audio range (about 48Khz = 48000 samples per second).


You want to just fit a curve to the data. Assuming the sampling rate is sufficient, a third-order polynomial would be plenty. At each point N, you fit a cubic polynomial to points N-1, N, N+1, and N+2, and then you have an analytic expression for the data values between those points. Shift over one, and repeat. You can average the values for multiple successive curves, if you want.

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