It could be similiar to what happens when uncompressed music is turned into an mp3, where intelligent judgement calls based on statistics are made wholly dependent on the trend of the waveform.Ingonator wrote:I don't think there is any "table" involved. First the responses used from waveforms of different synth also sound different and second this is not limited to "analog" waveforms. If you use a complex waveform you'll also receive a complex filter. Anyway if the waveform contains a part that corresponds to a resonant waveform the result could be that you receive some kind of resonant sound too.goldenanalog wrote:If you were to make assumptions about the waveform that you were analysing, you could manufacture a filter based on assumed changes in the waveforms' spectra *post* filter. You're absolutely right, of course: the previous statement implies that probability is used like using Calculus to calculate the slope value at discrete points of an assumed *known* function.pdxindy wrote:Ingonator wrote: It does not totally emulate a filter but by loading a single cycle wave you get the "impulse response" of a filter.
Ingo
That makes no sense at all. You cannot possibly determine what a filter does unless you analyze 2 waveforms, one pre filter and one post filter to determine how a filter affected the original waveform.
My guess is that Tone2 looked at enough filter data to build filter models into a table that are recalled and fitted to the waveform being analyzed. What might be funner (If I've read it right) is have the filter response continously change over time as DSP is used to examine each zero-crossing periodic sample (as an example) and update the filter response to that sample.
If you mix two waveforms within one oscillator you also get a kind of mixed filter.
Ingo
OK: Starting with reference 'parts' of filter types in a table-you examine a single-cycle waveform, and besides looking at it's spectra, you assume that:
1.) It is *post*-filter processed; and consequentially:
2.) It has an original shape determined by what template(s) the assumed-post-filtered waveform fits best. Then: looking at the waveform as a single point, your final filter shape covering the audio spectrum is a projection modelled upon statistically-based probabilities gained through intense scientific analysis of known synthesizer filter types.
Considering this method of filter-characteristic extrapolation/regeneration along with the current breed of desktops leads me to believe that it would be relatively easy to have quite an extensive table of filter 'pieces' resident to construct a complex filter from; and that the whole waveform analysis/resynthesis thing could happen practically in real time.
I may be completely wrong about applying this to Rayblaster; but it is one relatively simple way of how building a complex filter based on comparison analysis of a single waveform might work.
One thing is clear: We are getting further and further away from the original intent of modelling analog synthesizers and instruments in software; Rayblaster (and Iris) clearly leave the realm of what's possible using purely analog synthesis or digital playback of recorded instruments.
And the reason is simple: We now have enough CPU resources available that we can start to do things in real time that had to be done off-line in the past.
Edited for clarity.
