Good news: It’s allmost done!
Uploading a WAV file to the software works, the code automatically analyzes the WAV by autocorrelation to find repeating waveforms and suggests a list of best fitting sample lengths.
Here an example how that looks after i uploaded a sample of a church organ patch of another instrument that i sampled at 96000 samples per second.
After selecting one of the waves (by pressing the Convert button), this is converted (streched or shriked) to a single cycle waveform of 4096 samples (currently using linear interpolation).
The only open task is the lowpass filtering…
For that i want to invest some more brain, just to try/offer different options
- the hard and rigid cutoff (like we see it in the factory waveforms)
- alternative softer filter slopes that prevent the well known filter pre- and post-ringing at steep flanks