Every sample -- Nuno Santos
No dia 20/03/2015, às 18:51, Marco Lo Monaco <marco.lomon...@teletu.it> escreveu: > How often do you update the LFO? Every buffersize (32/64 samples)? > > M. > >> -----Messaggio originale----- >> Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp- >> boun...@music.columbia.edu] Per conto di Nuno Santos >> Inviato: venerdì 20 marzo 2015 19:06 >> A: A discussion list for music-related DSP >> Oggetto: Re: [music-dsp] Glitch/Alias free modulated delay >> >> Hi, >> >> Today I have used a piece of code which is on musicdsp for testing this out >> again. >> >> http://musicdsp.org/archive.php?classid=4#98 >> <http://musicdsp.org/archive.php?classid=4#98> >> >> I was able to have a delay changing in time without any kind of artefact or >> glitch. However I have only managed to get this results by changing the >> parameter by myself. >> >> When I say, manually moving the parameter by myself I say that I update a >> property which will be linearly interpolated in time (500ms). >> >> When I try to apply the modulation which is a value between -1 and 1, that >> comes from an LFO, I always end up with artefacts and noise >> >> I don’t understand why it works so well when I move the parameter value >> (which is also changing constantly due to interpolation, and it doesn’t work >> when I apply the modulation with the lo… >> >> Any ideas? >> >> This is my current code >> >> void IDelay::read(IAudioSample *output) >> { >> double t=double(_writeIndex)-_time; // works perfectly moving the >> handle manually with the value being interpolated before getting the >> variable _time; >> //double t=double(_writeIndex)-_time+_modulation*_modulationRange; >> >> // clip lookback buffer-bound >> if(t<0.0) >> t=_size+t; >> >> // compute interpolation left-floor >> int const index0=int(t); >> >> // compute interpolation right-floor >> int index_1=index0-1; >> int index1=index0+1; >> int index2=index0+2; >> >> // clip interp. buffer-bound >> if(index_1<0)index_1=_size-1; >> if(index1>=_size)index1=0; >> if(index2>=_size)index2=0; >> >> // get neighbourgh samples >> float const y_1= _buffer[index_1]; >> float const y0 = _buffer[index0]; >> float const y1 = _buffer[index1]; >> float const y2 = _buffer[index2]; >> >> // compute interpolation x >> float const x=(float)t-(float)index0; >> >> // calculate >> float const c0 = y0; >> float const c1 = 0.5f*(y1-y_1); >> float const c2 = y_1 - 2.5f*y0 + 2.0f*y1 - 0.5f*y2; >> float const c3 = 0.5f*(y2-y_1) + 1.5f*(y0-y1); >> >> *output=((c3*x+c2)*x+c1)*x+c0; >> } >>>> On 20 Mar 2015, at 14:20, Bjorn Roche <bj...@shimmeo.com >>> <mailto:bj...@shimmeo.com>> wrote: >>> >>> Interpolating the sample value is not sufficient to eliminate artifacts. >>> You also need to eliminate glitches that occur when jumping from one >>> time value to another. In other words: no matter how good your >>> sample-value interpolation is, you will still introduce artifacts when >>> changing the delay time. A steep low-pass filter going into the delay >>> line would be one way to solve this. (this is the idea of >>> "bandlimiting" alluded to earlier in this discussion.) >>> >>> I can say from experience that you absolutely must take this into >>> account, but, if memory serves (which it may not), the quality of >>> interpolation and filtering is not that important. I am pretty sure >>> I've written code to handle both cases using something super simple >>> and efficient like linear interpolation and it sounded surprisingly >>> good, which is to say everyone else on the project thought it sounded >>> great, and that was enough to consider it done on that particular project. >>> >>> HTH >>> >>> >>> >>> On Fri, Mar 20, 2015 at 6:43 AM, Steven Cook >>> <stevenpaulc...@tiscali.co.uk <mailto:stevenpaulc...@tiscali.co.uk>> >>> wrote: >>> >>>> >>>> Let suppose that I fix the errors In the algorithm. Is this >>>> sufficient >>>>> for a quality delay time >>>>> Modulation? Or will I need more advance technics? >>>> >>>> That's a matter of opinion :-) My opinion is that the hermite >>>> interpolation you're using here (I didn't check to see if it's >>>> implemented >>>> correctly!) is more than adequate for modulated delay effects like >>>> chorus - I suspect a lot of commercial effects have used linear >> interpolation. >>>> >>>> Steven Cook. >>>> >>>> >>>> >>>> -----Original Message----- From: Nuno Santos >>>>> Sent: Thursday, March 19, 2015 6:28 PM >>>>> To: A discussion list for music-related DSP >>>>> Subject: Re: [music-dsp] Glitch/Alias free modulated delay >>>>> >>>>> Hi, >>>>> >>>>> Thanks for your replies. >>>>> >>>>> What I hear is definitely related with the modulation. The artefacts >>>>> are audible every time the modulation is applied: manually or >>>>> automatically (please not that I have an interpolator for manual >>>>> parameter changes to avoid abrupt changes). I think I was already >>>>> applying an Hermit interpolation. This is my delay read function. >>>>> >>>>> void IDelay::read(IAudioSample *output) { float t = >>>>> _time+_modulation*_modulationRange; >>>>> >>>>> if (t>(_size-1)) >>>>> t=_size-1; >>>>> >>>>> float sf; >>>>> >>>>> #if 0 >>>>> sf = _buffer[int(readIndex(t,0))]; >>>>> #else >>>>> float const y_1= _buffer[int(readIndex(t,-1))]; float const y0 = >>>>> _buffer[int(readIndex(t,0))]; float const y1 = >>>>> _buffer[int(readIndex(t,1))]; float const y2 = >>>>> _buffer[int(readIndex(t,2))]; float const >>>>> x=readIndex(t,0)-int(readIndex(t,0)); >>>>> float const c0 = y0; >>>>> float const c1 = 0.5f*(y1-y_1); >>>>> float const c2 = y_1 - 2.5f*y0 + 2.0f*y1 - 0.5f*y2; float const c3 >>>>> = 0.5f*(y2-y_1) + 1.5f*(y0-y1); >>>>> >>>>> sf=((c3*x+c2)*x+c1)*x+c0; >>>>> #endif >>>>> >>>>> *output = sf; >>>>> } >>>>> >>>>> float IDelay::readIndex(float t, int offset) { float >>>>> index=_writeIndex-t+offset; >>>>> >>>>> if (index<0) >>>>> index+=_size; >>>>> >>>>> return index; >>>>> } >>>>> >>>>> Thanks, >>>>> >>>>> Regards, >>>>> >>>>> Nuno >>>>> >>>>> On 19 Mar 2015, at 18:12, David Olofson <da...@olofson.net >> <mailto:da...@olofson.net>> wrote: >>>>>> >>>>>> On Thu, Mar 19, 2015 at 6:15 PM, Nuno Santos >>>>>> <nunosan...@imaginando.pt <mailto:nunosan...@imaginando.pt>> >>>>>> wrote: >>>>>> [...] >>>>>> >>>>>>> If I use interpolation for buffer access I experience less glitch >>>>>>> and more alias. >>>>>> >>>>>> What type of interpolation are you using? I would think you need >>>>>> something better than linear interpolation for this. I'd try Hermite. >>>>>> That should be sufficient for "slow" modulation, although >>>>>> theoretically, you *should* bandlimit the signal as soon as you >>>>>> play it back faster than the original sample rate. >>>>>> >>>>>> For more extreme effects (which effectively means you're sometimes >>>>>> playing back audio at a substantially higher sample rate than that >>>>>> of your audio stream), you may need a proper bandlimited resampler. >>>>>> (Apply a brickwall filter before the interpolation, and/or >>>>>> oversample the interpolator.) >>>>>> >>>>>> >>>>>> -- >>>>>> //David Olofson - Consultant, Developer, Artist, Open Source >>>>>> Advocate >>>>>> >>>>>> .--- Games, examples, libraries, scripting, sound, music, graphics ---. >>>>>> | http://consulting.olofson.net <http://consulting.olofson.net/> >> http://olofsonarcade.com <http://olofsonarcade.com/> | >>>>>> '---------------------------------------------------------------------' >>>>>> -- >>>>>> 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/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/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 >>>> -- >>>> 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/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 >>> >>> >>> >>> -- >>> Bjorn Roche >>> @shimmeoapp >>> -- >>> 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/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 > > -- > 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
