What I've been looking at might be obvious to a lot of you, but it was
interesting to me. Without using any plugins at all and no aux/bus
channels, etc., there seems to be an approximate 1.94ms delay (93 samples @
24 bit 48kHz) every time a new audio track is recorded next to pre-existing
tracks.

This seems to be almost entirely independent of the Playback Engine's
Samples and DAE Buffer Levels. I experimented with the following settings:

512 Samples, DAE Buffer Level 2
128 Samples, DAE Buffer Level 2
128 Samples, DAE Buffer Level 4
1024 Samples, DAE Buffer Level 4
64 Samples, DAE Buffer Level 0

None of them seemed to affect my results except in the last case where it
brought the delay down to about 91 samples instead of 93.


This was tested with ProTools LE 6.2.3 running on a dual processor G5 w/ 2mb
RAM.

I started with a short 4 second file inside a new, empty 24bit, 48kHz
session. I created 6 empty mono tracks, and slided the file into the start
of track 1. I then played it to output 1, which was plugged directly into
input 5 on the 002R, recording that directly into track 2. I muted tracks
appropriately to avoid feedback and moved on down the list, 1 at a time, so
after the first pass of recording to track 2, I then used track 2's output
with its newly created file to record it into track 3, and so on down. Each
time there's a 93-94 sample delay, so after 5 passes, it adds up to almost
10ms.

In the real world, these subsequent passes would obviously be overdubs like
vocals, guitars, etc., and you would presumably be monitoring with a roughly
stable mix while doing overdubs (rather than the explicit/cumulative example
I set up), so the problem might not be so pronounced. Still, it seems to me
that this could cause a smearing of audio tracks, especially in mixes
involving a lot of overdubs, or at least certain type of overdubs more
susceptible to problems caused by short delays, like percussion hits...

Am I crazy? This has nothing to do with the "monitoring latency" that
people are always talking about - which has its workarounds. This has a
workaround too, if you're willing to go shifting new tracks back by 93
samples every time you record additional audio!

Why doesn't DigiDesign program some sort of offsetting fix into their
software?

Why doesn't anyone talk about this problem? Am I just missing something
here?