I'm mixing a project on a Digi 002 at a studio Wednesday that has some nice outboard eq's(api) and comps(summit) Should I tinker with the obg? Or stay in the box? If I simply use the EQs as an insert i will be dealing with latency issues correct?

Yes you will, though...just keep the buffer size small ex.128, and you should be able to monitor without much distraction. Once you are satisfied and done with the tweaking of the sound, print it and slide it forward the amount of samples that was delayed. Hope it helps.........Joaquin.

If you're using the converters in the 002, make sure the benefit of the outboard vs. inboard is enough to warrant the extra digital to analog and analog back to digital conversions.

I don't use the converters in my 002, and they do get a bit of 'bad press' around here, but they're totally workable. Not really the smoothest nor the most musical, but totally workable.

The difference between good outboard (the gear you mention certainly comes under that category) and plugins is pretty immense, in my opinion.

Here's a question I've had (now that we're talking about it)

Does PT/002Rack compensate for the latency between the various inputs like, native A/Ds, SPDIF and ADAT. So if youre recording 18 inputs at once, will they all be aligned?

Same question for the outputs...are they compensated?

I'm doing things outboard/ITB and I'm noticing a delay on percussion when i send and return to an external device. I just want to make sure that when i finally get the 8 A/D/A and a console, I won't have my ADAT stuff coming out before the native D/As and the SPDIF.

Cool thanks guys..yea it may be worth the time to go through the print process if we are talking the diff between api 560's and 550's as opposed to digi rack eq's. Although I hear the digirack eq is pretty good.

thumbsup

I meant the new digi rack eq is supposed to be ok.

Men...That's a great question!!? I just do not know the answer, and as far as I know, there's no mention of it in the reference manual...
I already recorded a bunch of music using both inputs...ADAT and Line...never had a problem and to tell you the true, I think that, acording to the distances, there's a diferent arrival time for each mic, and each instrument has it's own timing...I just find it pretty hard to test...with a splitter!? actually., I have an ISA428 wich give me simultaneus output, ADAT and analog...I'll do the experiment and see how far apart the waves are. Any way is very valid question and would be a major issue for what I had contemplated as a very reliable traking system!!
I'll post back...maybe someone else knows the answer!?

I am scared to even know the results of that test.

My dear friends...27 samples later than the line input!!....just one more workaround for my LE system

Thats not bad at all, I was expecting worse.

Better late than never, I suppose.

Couldn't this just as easily be a latency issue at the ISA's A/D as opposed to a discepency between the 002's Analog and ADAT inputs? Seems like it would be impossible to automatically compensate for another devices unknown latency if that's the case.

wow for now I'm gonna pretend I didn't here this.

Aw crap I can't bury my head in the sand!
Hey Jaoquin, what did you use as the source to get into the lightpipe inputs of the 002?
I'm going to do a test too and see what I find. Dod I guess you can't take anything for granted these days. This is could cause a hammer to fall upon my 002.

Yhea I think so. I believe that this latency is going to exist regardless of the system you are recording to...LE or HD, for is only "Input" and the conversion is being made exteriorly...not much to compensate for!?
As stated, I used a Focusrite ISA428 as the master clock (another issue??!!) recording at 48KHz 24bit.
I'm not going to think too much about this stuff. Probably, next time I record more than 8Trks, I'm going to nudge the ADAT Trks, by the re-mesured amount of samples, and hear if there's any Improvement...I doubt it though, for I'd have all my Phase already check acounting for any posible latency!?...I guess we can spend some time thinking about this stuff
Cheers.......................Joaquin.

Hey Joaquin...

thanks for doing this test... i don't think 27 samples is TOOO bad.. but maybe a little sliding is in order.

I guess the audio is going to come out at a different time as well.... Could be difficult to test..

bb

just remember a 1 sample disparity is enough to cause comb-filtering