On 2013-05-21, Dave Hunt wrote:
So, there must be quite a lot going on in Focusrite's Liquid Channel.
http://global.focusrite.com/mic-pres-channel-strips/liquid-channel
Reputedly Focusrite license a system from Sintefex.
http://www.sintefex.com/docs/appnotes/dynaconv.PDF
There's altogether too much hype there. Yes, you can do what they do:
characterize the short term near-LTI part of the system in steady state
and then brute force apply that sample by sample in a time-variant
convolution. That is a pretty powerful operation, but it's not really
the form you need for compressors. I believe it's both underkill and
overkill at the same time. (The input portion seems legit as far as I
can tell, presuming they implemented it right.)
That's because typical analog compressors are single band, which means
they operate on instantaneous amplitude only. There's filtering
complexity to be sure, but it's entirely in the side chain, whereas the
main path is more or less just a voltage controlled amplifier. The
Focusrite architecture gets it the other way around: a brutal amount of
brute force processing power is being used for an operation which
essentially ends up recreating a constant, more or less unity, EQ curve,
while the side chain isn't being modelled at all beyond instantaneous
nonlinearity.
In analog compressors, and especially the better ones, the side chain
which determines the eventual gain in the main one is exceedingly
carefully tuned, stateful, and at longer time scales surprisingly
nonlinear. It has different attack and decay time constants, it can
employ slew rate limited ramps purposely, and sometimes it even has
differently EQ's subbands with different constants. That is not
something an architecture like Focusrite's can capture, not in analysis
nor in synthesis. Mostly you're not going to notice it, of course, but
a suitable mixture of steady state background, transients and silence
will almost certainly show a definite difference to the original system
being modelled.
Additionally they say in the second link that they interpolate impulse
responses linearly. That is a bad idea in itself, because it'll almost
always lead to passband ripple between the endpoints, and if you're
heavily into transient content like me, intermediate forms with
time-variant allpass terms, muddying up the temporal structure of the
signal. Combining the simulation of the nonlinear preamp and the
compressor into a single, simple circuit like this buys them easy
analysis, but it also makes their synthesis side unsuited to the task at
hand and so nasty to analyze properly they don't even try it but resort
to passing ad hoc intuitions to it.
That sort of thing is Unclean. It will get the macroscopic stateless
nonlinearity of the preamp more or less right, in steady state, for
sparse quasiperiodic LF signals, the overall EQ curve more or less in
the ballpark, and it'll capture the compression characteristic for
slowly and smoothly varying envelopes. But it'll definitely not be a
"precise" replica of any and all analog input stages. In fact I'm pretty
sure you can even hear alias in the output, because when you do it the
way they claim to, that first crucial coefficient of the impulse
response, as a function of the input signal, will constitute an
arbitrary table lookup/waveshaper of very high polynomial order. That
sort of thing is very easy to drive into audible aliasing unless they
employ truly exorbitant oversampling rates in the intermediate
stages...which you can't really do without running into processing power
and latency constraints.
--
Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front
+358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
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