Delay compensation for going D/A/D

[bplr]

Tried to find this question via the search, and didn't exactly what I'm looking for. Hope I'm not repeating.

So, I'm trying to mix more and more with outboard gear. How do you guys deal with the delay created by (I'm assuming) your converters? I'm on a Digi002 and I get a pretty serious lag after going D-A and A-D. For auditioning settings I've had to send everything but the track I'm sending out to a separate aux with a fully wet delay and push the time till it sounds right. Once I get it, I bounce it and line the little bitch up by sight (sigh).

It seems like there should be an automatic calculation that PT could do and apply it if it knew which tracks were going out and back. A simpler solution would be creating a reverse delay plug that you could just slap on the incoming track. PT would then play the incoming track 60ms (or whatever) later, and I wouldn't have to send every goddam track but one to a new aux ...

Any thoughts? What am I missing here?

Justin

[MoreSpaceEcho]

if i read your post right you're saying you run, say, a snare track out of the daw to an Analog Whatever, and record it back in as a new track, yeah? AFAIK there's no way to get around the latency. there's just an inherent delay in the conversion. if i'm wrong that'd be great...

[bplr]

well, i WAS asking, "how do you go about auditioning settings when you have to listen to the drums 65ms late?" i realize that while running a signal through D/A/D in real time is going to cause unavoidable latency to some degree (varying with your gear/budget), and i'd still love to hear opinions on that ...

howEVER, i just went back into PTLE to run a test example and see what the exact latency was, and something crazy happened. it looks like PT is correcting for it once the audio gets recorded to a new track.

here's what's going on: i ran a guitar track out of the 002 line outs into an outboard pre/comp and back in on a line in. when i armed the track and listened to both the original and compressed together, there was a slap delay. when i record that track and zoom all the way in on the first transient, it lines up perfectly. playback of both tracks together gives me no slap once it has been recorded, and it appears that immediately after recording the D/A/D conversion, PT is lining it up for me.

if that's what it happening, how does it know? i just updated to 7.3.1 - is this something with 7.3? i'm confused because i know i used to have to line them up after recording. i can't find anything on Digi's site, the DUC, Google, or here.

FURTHERmore, i tested a 20Hz square wave (generated by PT) the same way, so i could see a precise transient. out of curiosity, i did a send from the receiving track to a bus and recorded that onto a 3rd track. the end result was:

track 1: original 20Hz square wave
track 2: D/A/D version, perfectly aligned
track 3: bussed from track 2 and recorded, 3166 samples late

???? Here's a picture -

http://www.bplr.com/Temp/Picture-2.jpg

of course, there's also the question about the shape of the waveforms, but i'm going to start another post for that one.

[Mane1234]

I ran into this a while back ago while busing some synth drum tracks out to an Aphex Big Bottom and then recording a new track. I'd have to slide it back a bit every time. I guess unless you can run everything through a board and just use the board inserts then it just has to be dealt with. I've never noticed Pro Tools trying to compensate for it on the way back in. I'll have to try it again. I never would have guess it would be as much as 60 ms.

[dirk_v]

LE, arcane as it is, still makes you compensate for the delay manually. It's such a PITA that I've found myself avoiding working methods that require me to do this, but here is: Create a new stereo Aux track and insert an instance of TimeAdjuster set to the required number of samples (in your case, 3166). Route the output of all your in-the-box tracks thru this aux track (quick and easy if you hold down Alt/Option while selecting the output path in the mixer). Create another stereo aux track with nothing on it and route your out-of-the-box tracks to it. Even better, you could just have the return from your outboard piped straight into this aux track for monitoring. Since all your other tracks are going through the TimeAdjuster path, they'll be delayed the required amount and you can hear everything in time.

For the times I do need to do this, it's usually just a matter of 'set and forget,' where I create the 'time delay' aux, set it to unity, hide it in the mixer, and try to forget it's there. As a handy aside, you can CTRL-click in the small text area below each fader in the mixer (say where it usually says "vol -4.5" or whatever and it will report the track's delay ("dly") in samples if you have any plugins or whatever inserted.

Best of luck.

-dv

[slowblue]

Sorry for stealing the thread but does the delay created by a plug in depend on how much processing the plug in requires? For example if on my 4 drum tracks I eq and compress the snare and kick quite heavily should I place the same eq and compression plug ins on the OHs to keep the delay constant across all the drum tracks or does the amount of processing on the kick and snare eq/compression cause even more delay as compared to plug ins that aren't doing anything (no eq adjustment, no compression). Thanks.

[earth tones]

Hey Justin...I feel your pain on this. It certainly is difficult to monitor and adjust an external device and accurately determine how it sounds in the mix, since after recording it you have to adjust for the 002's D/A and then the 002's A/D (and if outboard device is digital, it's D/A and A/D). For something like an outboard delay or reverb it probably would not require delay compensation as long as it sounds right to you when monitoring the effect live, if you are running those time-based effects 100% wet. When you aren't running them 100% wet, then the dry signals will probably exhibit comb filtering if not panned hard L and R, I assume.

I guess this is simply the price you pay for living on the lower end of the digital/analog-hybrid home studio spectrum. You simply have to come up with a suitable workflow to accommodate these realities.

I have never seen LE automatically compensate for conversion delays when sending to outboard. That is interesting.

[bplr]

I guess this is simply the price you pay for living on the lower end of the digital/analog-hybrid home studio spectrum. You simply have to come up with a suitable workflow to accommodate these realities.

i know. sucks. but not a huge deal. i'm adapting ...

I have never seen LE automatically compensate for conversion delays when sending to outboard. That is interesting.

none the less, i can't seem to explain this any other way. i gotta believe from what i'm seeing that i'm actually seeing PT slide it back into place.

[earth tones]

I guess this is simply the price you pay for living on the lower end of the digital/analog-hybrid home studio spectrum.

Sorry, the way I worded that statement was accidentally accusational. It was absolutely meant to say "I guess this is simply the price WE pay....", because I am right there with you.

Just wanted to clarify that typo.

[joel hamilton]

Isnt it in 7?

"compensate for delay after record pass" or something like that?

I thought I saw it in the I/O setup window.... in TDM though.

Not the insert or H/W delay, but an actual check box for compensating after record....

Check in the I/O setup dialog, see if you have something there now that wasnt there before.

[spacelabstudio]

It seems like there should be an automatic calculation that PT could do and apply it if it knew which tracks were going out and back. A simpler solution would be creating a reverse delay plug that you could just slap on the incoming track. PT would then play the incoming track 60ms (or whatever) later, and I wouldn't have to send every goddam track but one to a new aux ...

Any thoughts? What am I missing here?

I know this probably isn't what you want to hear, but Cubase just does this. You create an external effect, assign inputs/outputs, and then you can have it measure the delay and compensate automatically for you.

Without the DAW doing it for you, seems like the easiest way is to make a copy of the track you want to apply the effect to, apply the effect to one copy and not the other. Playing it back, you should hear some comb filtering, like a flanger stuck in one position. Play with the amount of delay until you make the phasiness go away. Done.

chris