BTW, I do know that it was developed with the Sonic Core SCOPE SDK, and I suspect it’s just using fairly routine DSP blocks, with a lot of care in tweaking the sound. (It runs in my mind that I might have seen some block diagrams on a forum back when he was developing it—the point is that I don’t think there is any cutting-edge tech involved.) But that’s all I know.
On Jun 18, 2014, at 10:40 AM, Nigel Redmon <earle...@earlevel.com> wrote: > No, Marco, sorry. I wish I did. The result is very good, and a huge leap from > the sim in the CX-3. Very annoying that they don’t support MIDI switching of > the speed… > > http://www.earlevel.com/main/2013/02/16/ventilator-adapter-in-a-mint-tin/ > > > On Jun 18, 2014, at 10:11 AM, Marco Lo Monaco <marco.lomon...@teletu.it> > wrote: > >> Ciao Nigel, talking about the VENTILATOR, do you know something more about >> its secrets/internals? >> :) >> >> M. >> >>> -----Messaggio originale----- >>> Da: music-dsp-boun...@music.columbia.edu [mailto:music-dsp- >>> boun...@music.columbia.edu] Per conto di Nigel Redmon >>> Inviato: mercoledì 18 giugno 2014 08:22 >>> A: A discussion list for music-related DSP >>> Oggetto: Re: [music-dsp] Simulating Valve Amps >>> >>> Well, some people think it’s close enough for rock n rock (amp sims), >> others >>> don’t. It’s the same with analog synths and virtual analog. But there’s >> also >>> the comfort of tube amps, and there’s the comfort of the limited sound >>> palette of using the amp that you know and love. Amp sims are really about >>> variety (can’t afford a Plexi, a Twin Reverb, SLO, AC-30, and a few >> boutique >>> amps? Now you can). >>> >>> I appreciate the old stuff, but I appreciate the convenience and >> flexibility of >>> the new stuff—to me it *is* close enough for rock n roll (new stuff in >>> general—I don’t play much guitar). I play a B3 clone because I hauled a >>> Hammond decades ago, and I hauled and still have (needs work) a Leslie, >> but >>> I’d just as soon use my Ventilator pedal on the CX-3 (yes, with >> programmable >>> leakiness and aging of the tone wheels, etc.)—more convenient, and gets >>> the sound I want. Others would shudder at the thought. Well, until their >>> backs start giving out…I know a hardcore, old-time B3 blues player >> (“Mule”— >>> a nickname he earning for hauling around his B3 and Leslies) who picked up >> a >>> clone for his aging back after hearing my CX-3 though the Ventilator. Not >> for >>> all gigs, mind you, but as an option to go with for some gigs. The point >> is that >>> if the tradeoffs are attractive enough, it’s easier to let yourself try >> new things >>> even if you feel that it falls ever so slightly short of what you’re used >> to, or >>> strays from your comfort zone. >>> >>> So while some might feel that amp sims haven’t arrived yet, other might >> feel, >>> “where the heck have you been the past decade?" ;-) >>> >>> >>> On Jun 17, 2014, at 6:59 PM, robert bristow-johnson >>> <r...@audioimagination.com> wrote: >>> >>>> On 6/17/14 8:24 PM, Nigel Redmon wrote: >>>>> (Thinking outside the nest…) >>>>> >>>>>> (...maybe that means opening up the LPF as the gain knob setting is >>>>>> reduced) >>>>> Yes >>>>> >>>>> And good discussion elsewhere in there, thanks Robert. >>>>> >>>> yer welcome, i guess. >>>> >>>> you may be thinking outside the nest; i'm just thinking out loud. >>>> >>>> i think, like a multieffects box, we oughta be able to simulate all >> these amps >>> (don't forget the Mesa Boogie) and their different settings in a single >> DSP >>> box with enough MIPS and a lotta oversampling. dunno if simulating the >>> 50/60 Hz hum and shot noise would be good or not (i know of a B3 emulation >>> that simulates the "din" of all 60-whatever keys leaking into the mix even >>> when they're all key-up). but they oughta be able to model each >>> deterministic thing: the power supply sag, changing bias points, >> hysteresis in >>> transformers, capacitance in feedback around a non-linear element (might >>> use Euler's forward differences in doing that), whatever. whatever it is, >> if >>> you take out the hum and shot noise, it's a deterministic function of >> solely >>> the guitar input and the knob settings, and if we can land human beings on >>> the moon, we oughta be able to figure out what that deterministic function >>> is. for each amp model. it shouldn't be more mystical than that (but >> there >>> *is* a sorta mysticism with musicians about this old analog gear that we >> just >>> cannot adequately mimic). >>>> >>>> and thanks to you, Nigel. >>>> >>>> L8r, >>>> >>>> r b-j >>>> >>>> >>>>> On Jun 17, 2014, at 4:07 PM, robert bristow- >>> johnson<r...@audioimagination.com> wrote: >>>>>> On 6/17/14 3:30 PM, Nigel Redmon wrote: >>>>>>> This is getting…nesty... >>>>>> yah 'vell, vot 'r ya gonna do? :-) >>>>>> >>>>>>> On Jun 17, 2014, at 10:42 AM, robert bristow- >>> johnson<r...@audioimagination.com> wrote: >>>>>>> >>>>>>>> On 6/17/14 12:57 PM, Nigel Redmon wrote: >>>>>>>>> On Jun 17, 2014, at 9:09 AM, robert bristow- >>> johnson<r...@audioimagination.com> wrote: >>>>>>>>> >>>>>>>>>> On 6/17/14 5:30 AM, Nigel Redmon wrote: >>>>>>>>>> >>>>>>>> ... >>>>>>>>>>> Anyway, just keep in mind that the particular classic amps >>>>>>>>>>> don’t sound "better" simply because they are analog. They sound >>>>>>>>>>> better because over the decades they’ve been around, they >>>>>>>>>>> survived—because they do sound good. There are plenty of awful >>>>>>>>>>> sounding analog guitar amps (and compressors, and preamps, >>>>>>>>>>> and…) that didn’t last because they didn’t sound particularly >>>>>>>>>>> good. Then, the modeling amp has the disadvantage that they are >>>>>>>>>>> usually employed to recreate a classic amp exactly. So the best >>>>>>>>>>> they can do is break even in sound, then win in versatility. >>>>>>>>>>> And an AC-30 or Matchless preset on a modeler that doesn’t >>>>>>>>>>> sound exactly like the amp it models loses automatically—even >>>>>>>>>>> if it sounds better— because it failed to hit the target. (And >>>>>>>>>>> it doesn’t helped that amps of the same model don’t necessarily >>>>>>>>>>> sound the same. At Line 6, we would borrow a coveted amp—one >>>>>>>>>>> that belonged to a major artist and was highly regarded, for >>>>>>>>>>> instance, or one that was rented out for sessions because it >>>>>>>>>>> was known to sound awesome.) >>>>>>>>>> what did you guys do with the amps when you borrowed/rented >>> them? was your analysis jig just input/output, or did you put a few high- >>> impedance taps inside at strategic places and record those signals >>> simultaneously? >>>>>>>>> Yes. For instance, sweeping the EQ with incremental settings >>> changes. >>>>>>>>> >>>>>>>> yes, another issue (which i didn't really touch on) is mapping the >>> settings of the knob to the internal (to the DSP) coefficients and >> threshold >>> values and such. that is "coefficient cooking" and is the same issue as >>> defining Q in EQs so that the knob behaves like the ol' Pultec or >> whatever. >>> your digital implementation might work very well, but if the position of >> the >>> knob in the emulation is not nearly the same as it was for the venerable >> old >>> gear (to get the same sound), someone might complain. >>>>>>> Oh yes, they *will* complain ;-) >>>>>>> >>>>>>>>>>> On Tue, Jun 17, 2014 at 6:58 PM, Nigel >>> Redmon<earle...@earlevel.com> wrote: >>>>>>>>>>>> On Jun 16, 2014, at 7:51 PM, robert bristow-johnson< >>>>>>>>>>>> r...@audioimagination.com> wrote: >>>>>>>>>>>>>> one thing that is hard to replicate is a sample rate that is >>>>>>>>>>>>>> infinity (which is how i understand continuous-time signals >>>>>>>>>>>>>> to be). but i don't think you should need to have such a >>>>>>>>>>>>>> high sample rate. one thing we know is that for *polynomial >>>>>>>>>>>>>> curves* (which are mathematical abstractions and maybe >>> have >>>>>>>>>>>>>> nothing to do with tube curves), that for a bandwidth of B >>>>>>>>>>>>>> in the input and a polynomial curve of order N, the highest >>>>>>>>>>>>>> generated frequency is N*B so the sample rate should be at >>> least (N+1)*B to prevent any of these generated images from aliasing down >>> to below the original B. >>>>>>>>>>>>>> if you can prevent that, you can filter out any of the >>>>>>>>>>>>>> aliased components and downsample to a sample rate >>> sufficient for B (which is at least 2*B). >>>>>>>>>>>>>> >>>>>>>>>>>>> This really goes out the window when you’re modeling amps, >>>>>>>>>>>>> though. The order of the polynomial is too high to implement >>>>>>>>>>>>> practically (that is, you won’t end up utilizing the >>>>>>>>>>>>> oversampling rate necessary to follow it), >>>>>>>>>> this is a curious statement *outside* of the case of hard >> clipping. >>> oversample by 4x and you can do a 7th-order polynomial curve and later >>> eliminate all of the aliasing. oversample by 8x and it's 15th-order. do >> *no* >>> oversampling and you can still make use of the fact that there's not a lot >>> above 5 kHz in a guitar and amp (so 48 kHz is sorta oversampled to begin >>> with). you can fit a quite curvy curve with a 7th-order polynomial. >>>>>>>>>> >>>>>>>>>>>>> so >>>>>>>>>>>>> you still be dealing with aliasing. Modern high gain amps >>>>>>>>>>>>> have huge gain >>>>>>>>>>>>> *after* saturation. In practical terms, you round into it >>>>>>>>>>>>> (with a polynomial, for instance), then just hard clip from >>>>>>>>>>>>> there on out, and there goes your polynomial (it can be >>>>>>>>>>>>> replaced by an approximation that's very high order, but what’s >>> the point). >>>>>>>>>> yes, we splice a constant function against a curve. if at the >> splice as >>> many possible derivatives are zero as possible, that splice appears pretty >>> seamless. this is why i had earlier (on this list) been plugging these >> curves: >>>>>>>>>> >>>>>>>>>> x >>>>>>>>>> f(x) = C * integral{ (1 - u^2)^M du } >>>>>>>>>> 0 >>>>>>>>>> >>>>>>>>>> (C gets adjusted so that f(1) = 1 and f(-1) = -1.) >>>>>>>>>> >>>>>>>>>> you can splice that to flat values at +/- 1 and the nature of the >>> function will not change appreciably from the polynomial in the region of >> the >>> splice. >>>>>>>>>> >>>>>>>>>> anyway, the whole point is to give the guys with golden ears no >>> cause to complain about hearing aliases. same with emulating sawtooths >> and >>> hard-sync synthesis. >>>>>>>>>> >>>>>>>>>>>>> Anyway, you pay your money, you make your choices. >>> Obviously >>>>>>>>>>>>> some really good musicians making really interesting music >>>>>>>>>>>>> use modeling amps. They don’t have to be better than tubes, >>>>>>>>>>>>> in order to be a win, just good enough to be worth all the >>>>>>>>>>>>> benefits. If you’re a session music, you can bring in the >>>>>>>>>>>>> truck with all of the kinds of amps that might be called on, >>>>>>>>>>>>> or you can bring a modeling amp, for instance. And going >>>>>>>>>>>>> direct into the PA or your recoding equipment…etc. I’m not >>> going to make judgments on what people should like, so I’ll leave it at >> that. >>>>>>>>>>>>> >>>>>>>>>>>>> One happy thing about the aliasing is that, given a decent >>>>>>>>>>>>> level of oversampling, it won’t be bad at lower overdrive >>>>>>>>>>>>> levels. At the higher the overdrive levels, the harder it is >>>>>>>>>>>>> to hear aliasing through all that harmonic distortion you’re >>> generating. So it could be worse... >>>>>>>>>> i really agree with this, Nigel. with *some* oversampling (but >>> theoretically not sufficient oversampling), you can get away with a lot >> (like >>> hard limits or whatever stuff goes on inside a transformer with core >> loss). i >>> would not say that you have to oversample to a ridiculously high degree >> just >>> because there is a hard-limit saturation in there or that your tube model >> is >>> not a polynomial approximation (but i wonder why you wouldn't try to fit >> the >>> grid-to-plate tube curve to a finite-order polynomial). >>>>>>>>> What I mean is... for a modern high-gain amp, the gain is on the >>> order of 2^16 (and the curve starts it’s significant bend up near 1). So >> most of >>> the signal, when you’re playing maxed out, is simply clipping hard. If >> your >>> goal is to not alias in the audio band at all, by figuring the max >> harmonic >>> component based on the order of the equivalent polynomial and the highest >>> freq of the guitar input coming in…well, your oversampling factor is going >> to >>> be a lot higher that you’re willing to implement. >>>>>>>> i understand. hard-hard-limit and you got harmonics going up to >>> infinity anyway. >>>>>>>> >>>>>>>>> There’s really no point in calculating a continuous polynomial >> over >>> that range that I can see. >>>>>>>> well, if it splices *well* to the clip region, it might *still* have >> a point. >>>>>>>> >>>>>>>>> It’s no big deal—I just brought it up because I often see people, >> here >>> and elsewhere, go down the thought path of... "OK, I want to make a guitar >>> distortion unit…if I keep my polynomial to order N, I only need to >> oversample >>> by (N+1)/2...", completely forgetting that when, in their code, they >> branch to >>> limit the output to +/- 1, their polynomial order just went out the >> window. >>>>>>>> yes, that's true (sorta). at least *if* the splice to the limited >> constant >>> value is not smooth. >>>>>>>> >>>>>>>> but you can make a polynomial match as many derivatives (equal to >>> zero) of the hard limit as possible (but that might be at cross-purposes >> to >>> getting the polynomial to follow a tube curve) and for levels that hit >> that limit >>> (so the code branches to the limit), if the overflow or spike isn't so >> bad, the >>> behavior isn't so far away from the "ideal" polynomial and the total >>> behavioral issue remains inside the window, i would think. >>>>>>> Yes, Robert…but, with the kind of gain necessary…OK, so you have the >>> y-xis as you output level, x-axis as input. To view the entire curve for a >>> Soldano Super Lead Overdrive, for instance, you draw the curve of your >>> choice to rise from y=0 and give you a soft bend into y=1 (full output). >> The >>> bend will be somewhere around x=1, ballpark (maybe it’s x=2 or 3, to allow >>> for lower input levels, but the point is that it’s a small number compared >> to >>> what’s coming next)…then you allow for x=30000 or so (a flatline from the >>> x=1..3 area). Is that not a pretty high order polynomial? >>>>>> well, yeah, and it might better be described as a function that is >>> discontinuous with most of its derivatives, even the 1st. >>>>>> >>>>>> so >>>>>> >>>>>>> The point being, yes the polynomial would be handy at low gain >>> settings, but you still need to build this thing to work at extreme gain >> settings >>> at the same time. >>>>>> okay, you mean with it cranked up so that it virtually hard limits. >> that's >>> not exactly what comes to mind about "warm" tube distortion. like those >>> DevilDrive guys (or was it the Kemper guys) built a 12AX7 preamp to model >>> (and i wonder how much that tells us about how a Fender Twin Reverb >>> cranked up to arcweld behaves like). >>>>>> >>>>>> but this is hard clipping distortion, not zero-crossing distortion, >> right? in >>> between the nasty hard limits, you might be able to decently model the >> tube >>> curves with finite-order polynomials. specifically the mapping curve from >>> biased grid voltage to biased plate voltage given a specific load line >> (which >>> may be affected by power sag). maybe you can cover that quite well with a >>> finite-order polynomial and emulate that with a finite sampling rate. but >> if it >>> clips, might be nasty, regarding aliases. >>>>>> >>>>>> the only thing i know how to tame down a hard limit (and it may very >>> well not be compatible with the characteristic tube curve) is to set as >> many >>> derivatives as possible to zero and splice the hard limit to that thing. >>> continuity to the 2M-th derivative including the hard limit. >>>>>> >>>>>>> So anything at the low gain settings is pretty insignificant for >> something >>> designed to handle the high gain settings. >>>>>> well, we gotta think sorta like the string theorists. we gotta >> imagine how >>> to seamlessly glue together two ostensibly incompatible systems. like how >>> do we crossfade from the low-gain behavior (the "warm tube sound") to the >>> behavior we like when it's cranked up to arc-weld? >>>>>> >>>>>> >>>>>>> Hence my feeling that there not much point to calculating how much >>> headroom you have—you can pretty much count on infinity. There may be >>> some reasons to do it—I’m not demanding that I have the right idea, just >>> simply explaining what I meant by my comments. In reality, it’s not so >> clear >>> cut, because as I mention before, the more you get into a situation where >>> aliasing will be big, at the same time you are in a situation where you’ll >> have >>> more generated harmonics to mask the aliasing. In the end, aliasing is >>> *mainly* a problem if you bend a guitar note and you heard harmonics going >>> in the wrong direction. For some reason guitarists just can’t get around >> that >>> (lol). >>>>>>> >>>>>>>>> BTW, the more the overdrive, the less the weaker upper harmonics >>> of your guitar matter, so you can cheat by rolling them off as you >> increase >>> drive. >>>>>>>> a useful idea. more pre-LPF as the grunge gets cranked up. >>>>>>>> >>>>>>>>> But you can’t rely on that too much, because guitar players like >>>>>>>>> to hang analog distortion stomp boxes in front of your modeling >>>>>>>>> amp, giving you powerful higher harmonics. :-) >>>>>>>> yeah, but can't you *still* pre-LPF that signal (the output of the >>> distortion stomp box) as the amp drive is cranked up? i dunno. >>>>>>> Yes, it’s definitely one place where you can win, and help yourself >>>>>>> make the best of a practical amount of frequency headroom. Probably >>>>>>> the biggest difference (between assuming direct, clean guitar >>>>>>> strings as input, and one that’s be pre-crunchified with a >>>>>>> stompbox) is that for the former you might get by with a >>>>>>> lower-order filter, because guitar string harmonics drops of >>>>>>> pretty quickly by themselves. (So, you might design an amp sim that >>>>>>> seems relatively alias-free, then get a customer or beta tester >>>>>>> complaining about the aliasing, and that's were you find out that >>>>>>> guitarists will still want to run their stuff into your sim, even >>>>>>> if you give them those functions in DSP.) >>>>>> well, i know there can be different specs. but for a 32-tap FIR LPF, >> you >>> can put the same brick-wall LPF on both guitar (that might not need it as >> bad) >>> and the grunge box. it's just that for clean amp setting, you might hear >> the >>> difference between your straight-grunge pedal and the LPF'd one (and it's >>> less necessary, maybe that means opening up the LPF as the gain knob >>> setting is reduced). >>>>>> >>>>>> -- >>>>>> >>>>>> r b-j r...@audioimagination.com >>>>>> >>>>>> "Imagination is more important than knowledge." >>>> >>>> >>>> -- >>>> 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 > > -- > 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
