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latency questions

On the latency topic, here are my observations.

The cheap solution: Terratec Phase26 USB. 
Works perfect and stable with an output latency of 16ms and input latency 
5ms = global latency of 21ms.
Very seldomly clicks at 9ms/4ms = 13ms.
Unuseable at its lowest setting (4ms/3ms = 7ms).

The better solution: RME Digiface connected to Hammerfall CardBus.
works perfect and stable with 2ms/2ms = 4ms.
unuseable at its lowest setting (1ms/1ms = 2ms).

I havent' tried optimizing my system for using ultra low latency, as
suggested by RME. Still, 4ms is MUCH better than 21ms. You get what you pay

Please be advised that I did not measure the total latency of, say running
from the A/Ds to the ADAT interface into the Digiface into the Hammerfall
CardBus through the buffers through Live's I/O through several effects
through Live's I/O through the buffers through the Hammerfall into the
Digiface into the ADAT interface and out to the D/As. And this is bascially
what we're after if we are looking at latency. 

Per, you mentioned you always listen to your instrument's signal before
going through the laptop chain. Now my question: how do you make sure this
way that you are "in time" with what comes out of your laptop if you
actually record a loop or something? I'll try to explain what I mean.

Let's say we have three timing systems running at the same speed but with
different offsets.
1) the sound of your instrument (the input)
2) "the inside of your laptop" - the timing system of e.g. Mobius
3) what comes out of the speakers (the output)

I think it's safe to say that 3 is later than 2 is later than 1. Let's call
the respective delays d12 and d23.

Note that there are different possibilities in which timing system you
perceive your own playing. If you listen to the signal going through the
converters, through the laptop and then coming back out of the speakers, 
listen to it in system 3 - case A. If you monitor your 
signal, you listen to it in system 1 - case B.

Case A:
You listen to a loop playing back (in timing system 3) and play along with
it (in timing system 1). Everything you play reaches your ear some time
t=d12+d23 after you play it. (nb: this is not at all unusual for people 
playing old church organs, both from mechanical, pneumatical and acoustic
delays). Hence, you start playing everything a little earlier, namely by t.
So your playing arrives at 2 (in Mobius for an overdub) actually d23 before
you hear it, but that is ok because the loop happens in Mobius by 
d23 before you hear it, too. So your overdub is perfectly in sync with the
original loop.

Case B:
You listen to a loop playing back (in timing system 3) and play along with
it (in timing system 1). Everything you play reaches your ear instantly, so
you don't have to "correct" anything. So something you overdub onto your
loop takes the time d12 after you play it to get to Mobius, and Mobius
actually played something d23 ahead of what you heard when you played your
overdub - hence your overdub happens with a delay of t - which, with crappy
hardware can be more than 20ms. (something like 32T at moderate house

So while it's perfectly cool to put your "unprocessed" instrument signal
into the mix together with any non-looper effects you might be using (like
reverb or chorus, which usually even have a predelay of their own, which 
can then shorten accordingly), I would recommend against doing so when
actually overdubbing loops.