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:
>> Well…yes, aliasing is the main issue that separates the digital world from
>> analog when it comes to amp modeling, but no, I don’t think it’s the main
>> issue in simulating a good amp :-)
>>
>> There are a lot of details in simulating classic amps—the controls of the
>> passive filters interact, etc.
>
> and if they stay linear, you oughta be able to lump some of these passive
> filters, no? if some component of the passive filter goes non-linear, you
> have to first decide if the nonlinearity is salient, and if it is, then model
> it.
My point was that if you change the bass, the midrange changes too, etc.—if you
want to model a classic amp, you have to not just get the sound right, but the
behavior of the amp (control interaction, the behavior when you tax the power
supply, etc.).
>> And you have to recreate the filtering before and after the “tube”—guitar
>> amps aren’t about flat frequency response.
>
> isn't that par-for-the-course, Nigel?
Yes it is. I just mentioned that for casual readers who aren’t very familiar
with guitar amps compared to tube hi-fi amps.
>> And the cabinets are a huge part of the sound.
>>
>
> *that*, and the loudspeakers themselves, is the hardest part, no? it's all
> three: 1. salient (so you can't ignore it), 2. non-linear, and 3.
> non-memoryless.
Another thing the cabinet (and I use the word in the sense that a guitar player
would—the cabinet with loudspeakers) brings is that the frequency of guitar
cabinets drops like a rock at a relatively low frequency (say 5kHz)—it can
allow some tradeoffs for efficiency.
>> Then there’s the approach that’s more like sampling (Kemper Profiling Amp),
>> a totally different direction—does it sound better?
>
> i don't understand how they're a totally different direction. i haven't seen
> nor heard these amps, i am merely reading the reference manual, but it
> appears they are doing the "Power Sagging", "Tube Shape", and "Tube Bias"
> emulation that, at least appears ostensibly the same direction as other
> modeling efforts.
Well, in one case you’re emulating the components of known guitar amps in DSP,
and the other you’re analyzing any amp, and essentially sampling its behavior.
>> 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.
>> 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. There’s really no point in calculating a
continuous polynomial over that range that I can see.
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 X, I only need to oversample by Y…”,
completely forgetting that when, in their code, they branch to limit the output
to +/-1, their polynomial order just went out the window.
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. 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. :-)
> --
>
> r b-j r...@audioimagination.com
>
> "Imagination is more important than knowledge."
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