Can you explain your notation a little bit? Is x[t] the sample index into
your signal? And t is time in samples?

I might formulate it as a Delta of indicies where a Delta of 1 is a normal
playback speed and you have some exponential rate. Would something like
this work?

delta *= rate
t += delta
y[n] = x[n - t + N]

My notation probably means something different from yours, but the idea is
the time varying index t accelerates or decelerates at a given rate. I've
kind of written a hybrid of pseudocode and DSP notation, sorry.

You would probably actually want some interpolation and for an application
like this one I would probably stick with linear interpolation (even though
most people on this list will probably disagree with me on that). Keep in
mind, though, that skipping samples might mean aliasing which will mean
low-pass filtering your signal (unless you know that there's no frequency
content in offensive ranges), and since you're essentially doing a variable
skip rate your low pass filter might either need to be aggressive or

Something about this algorithm scares me in it's seemingly unbounded need
for memory. Seems like a lot of heuristic constraints...


On Feb 6, 2018 06:45, "Maximiliano Estudies" <> wrote:

I am having trouble with this concept for quite some time now, I hope that
I can explain it well enough so you can understand what I mean.
I have signal stored in a buffer of known length. The buffer must be read
at a variable speed, and the variations in speed have to be exponential, so
that the resulting glissandi are (aurally) linear. In order to do that I
came up with the following formula:

x[t] = t * sample_rate * end_speed^(x[t] / T)     where T is the total
length of the buffer in samples.

This doesn’t seem to work and I can’t understand why.

And my second question is, how can I get the resulting length in
milliseconds? (how long will it take to play the whole buffer)

I hope I managed to be clear enough!


Maximiliano Estudies
+49 176 36784771 <+49%20176%2036784771>

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