That's exactly what I would also suggest! The linear interpolation is used in commercial products more than one could think.
> -----Messaggio originale----- > Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp- > boun...@music.columbia.edu] Per conto di Nigel Redmon > Inviato: sabato 21 marzo 2015 00:07 > A: A discussion list for music-related DSP > Oggetto: Re: [music-dsp] Glitch/Alias free modulated delay > > Suggestion: > > Make it work with linear interpolation first. > > The implementation is extremely simple—it won’t take much of your time to > try it—and you’ll eliminate most of the problems (buffer wrap, etc.) without > getting confused about whether your interpolation scheme is the fault. > > Plus, you’ll have a baseline to compare higher-order improvements with. > Linear interpolation sounds better than most people would guess, with > typical musical input (musical interments usually have weaker upper > harmonics), so you’ll have a better idea of whether you’re getting your > money’s worth with more elaborate methods. > > > > > On Mar 20, 2015, at 2:10 PM, Nuno Santos <nunosan...@imaginando.pt> > wrote: > > > > > >> On 20 Mar 2015, at 18:58, Alan Wolfe <alan.wo...@gmail.com> wrote: > >> > >> One thing to watch out for is to make sure you are not looking > >> backwards AND forwards in time, but only looking BACK in time. > > > > This is how I calculate the read index: > > > > float t=_writeIndex-_time-_modulation*_modulationRange; > > > > if(t<0) > > t+=_size; > > > >> > >> When you say you have an LFO going from -1 to 1 that makes me think > >> you might be going FORWARD in the buffer as well as backwards, which > >> would definitely cause audible problems. > > > > I have tried to rescale the LFO to fit between 0 and 1 and it doing the same > artefacts: > > > > > > // this where delay gets updated with lfo float lfo = > > (_lfo.step()-1.f)/2.f; > > > > delay.setModulation(lfo); > > > >> > >> your LFO really should go between -1 and 0, you then multiply that > >> value by the number of samples in your buffer (minus 1 if needed, > >> depending on your design and timing in your code), and then subtract > >> that value from your "write index" into the buffer, making sure to > >> handle the case of going negative, where your subtracted offset is > >> greater than your current write index. > > > > I even tried to change from > > > > _time+_modulation*_modulationRange > > > > to > > > > _time-_modulation*_modulationRange > > > > Exactly the same issues…. > > > > :/ > > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, dsp > links http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
