[LAU] 1980's cds: analog to digital conversion

[Monty Montgomery]

> Just two general-purpose first order IIR sections
> is all you need for either the forward or inverse
> filter. Any textbook on digital filters will tell
> you how to program them. Inverting the LF filter
> requires an extra pole below the audio range to
> avoid infinite DC gain.

I'm aware of how to construct digital IIR filters :-)  I was hoping
you had a URL to a nice official analog topology.  The specific
implementation details matter.

> The channel EQ you'll find on most digital mixers is
> not linear-phase at all, nor acausal.

OK.  Time to become incredibly overspecific:

Every digital EQ implementation I'm aware of for Linux is linear
phase.  I wrote a few of 'em.

One could just build a digital equivalent of any of the old analog
topologies.  For many filters (eg, compressors and the like) this is
totally the way to go.  For EQ, I'll take a linear phase
implementation any day.  That's the route taken by every piece of FOSS
EQ source code I've ever seen (it would not be surprising if I missed
a few).  If you say VST has done a few this way (for whatever reason)
then I believe you.

> In almost all
> cases it's just first and second order IIR filters.
> For plugins anything goes, but the most of them are
> not linear phase, and no filter operating in real
> time can ever be acausal - by definition.

Negative delays are perfectly possible in digital.  Well, if you
ignore the wallclock (assume a global system latency, and a local
negative latency within the system).  It's just a semantic/terminology
argument at this point.

Cheers,
Monty