Need to convert "virtual analog" waveform to digital

[Argitoth]

I think it would be much easier just to ask: Can Resampler in the WDL library be used for variable rate? I have an array of ratios (one that corresponds to every sample of the array holding the waveform) that I want to send it to change and interpolate it to a new array.



Red = my current waveform.
Black = the waveform I need to get to.

The analog waveform is currently held in a vector struct

{length, amplitude}

Each sample has its own length. You could say each sample has its own samplerate... 1/length = samplerate.

Let me explain how I arrived at the "red" (pseudo analog) waveform in the first place. The analog waveform was previously a digital one. I simply took each sample (amplitude value) and stored it in the struct {length, amplitude}. Originally, of course, each length of amplitude was a fixed value of 1/samplerate. Then I simply increased or decreased the length value for every given amplitude in the vector, a.k.a. "variable rate". That being said, maybe I went ahead of myself and I should be doing things differently altogether.

Any suggestions on what API or functions or methods or algorithms or whathaveyou that will help? I know WDL has a resampler... possibly a variable rate resampler? The end result should sound good, so at some point I need some kind of interpolation going on.
Thank you!

[kaklaudioparlour]

I don't know about Reasampler, but I imagine it could with some work.

You might want to consider converting your input data into a function that it continuos, so you can sample it like:

function getSample(x) {

  //Do interpolation here

  return y; 

}

and use interpolation (something better than linear) to fill in the gaps between sample points.

then you can loop over it to populate a fixed rate buffer:

for(var i; i<buffer_length;++i) {

    buffer[i] = getSample( i * timeStep );
}

If you want to go more deeply into it, the top answer here provides an algorithm to create a perfect reconstruction.

-kakl

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