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-   -   Protools low latency foldback (https://www.gearslutz.com/board/music-computers/107979-protools-low-latency-foldback.html)

brentnowlan 6th February 2007 01:30 AM

Protools low latency foldback
 
So heres the deal...I have a 002 rack and im wanting to use the 7-8 outputs for the foldback mix, but whenever you put a send on a track that is record enabled, it will bypasses the send when low latency mode is engaged, but if you turn off low latency mode it will work but with noticeable delay. Is there a way around this?

theblue1 6th February 2007 02:04 AM

Let me guess:

-- you want to put something like a reverb on a live track for headphone feed, something where the effect being delayed by 5 or 10 ms is no big deal but having the actual singer or guitarist delayed is unacceptable past the near zero latency monitoring offered by the 002?


I dunno because I'm not an 002 guy but it took me a few moments to understand the question. But consider this a vamp until the 002 folks show up... heh (And, actually, I believe this will work.)


I use (the somewhat parallel device) the MOTU 828mkII which also offers an onboard near zero latency mode.

On the MOTU, the NZL monitoring is engaged on the harward unit and the through-the-box monitoring with its way-not-near-zero latency (about 8 ms at 44.1 with 128 buffers) is controlled from my DAW software (in the input channel -- more or less where it is on PT LE, IIRC).

So to do what I think you want to do I would set up the "dry" sound in the "DSP CueMix" (near zero latency onboard digital mixer on the MOTU) and...

... oh yeah... then I'd insert the 'verb (echo, whatever) in the chosen input channel of my DAW, ENABLE input/source monitoring for that track and then set the reverb's output to EFFECT ONLY.



I was thinking (we do all know that's dangerous, yeah?) that the 002/PT LE relationship was similar -- but hopefully I've typed for so long someone who actually knows WTF they're talking about has shown up...

Good luck! heh

Benmrx 6th February 2007 04:25 PM

Why not just turn LLM off, and put your Hardware Buffer at 64.

Pulled this from a link in another thread

Quote:

If your HWB is 64, and you're monitoring through headphones, then the latency is the same as if you were monitoring a zero-latency mix through speakers and sitting with your ears 7 to 9 feet from the speaker cones. If you're monitoring through speakers, and sitting 3' away, HWB=64 increases latency to the equivalent of sitting 10-12' away. 7-12 feet is a pretty typical distance for musicians to sit from speakers in a control room. It's closer than they might be to each other on a large stage. It's much closer than the most distant members of a symphony orchestra are from each other.
I track straight into the 002 all the time, with verbs and such for singers and have never had issues with myself or clients regarding latency.

theblue1 6th February 2007 05:33 PM

Well, we're all different, aren't we?

Ask around and I think you'll find plenty of people who have problems with that level of latency under some circumstances.

I feel distinctly uncomfortable when I go D.I. with my Strat into my MOTU box's internal cuemix, which is NZL -- probably around 2 ms. It can be done but it definitely feels funny. (Yes, I definitely know that's the equivalent of an amp sitting about two feet away, in a sense; what can I tell you, it feels weird and I was totally NOT expecting it to be any kind of problem at all.)

And -- for me -- the 8 ms round trip running through the computer (to try to make use of, say, an amp sim plug) is simply undoable.

Different folks are different. If your clients don't bug out over that latency kind of latency, I think you're lucky. (I was thinking the monitoring latency on an 002 was less than that, closer to 5 ms but that's just a number that was salted away in my head. Still at 64 samples, I would think that's about the monitoring latency you'd have to expect -- barring the use of plugs that might unavoidably delay the audio further, anyhow.)


You say people don't notice something like an 8 ms delay -- but I discovered that my DAW was not compensating for conversion and tranport latency (as it relates to overdubs) when I cut a bongo part and thought, within my abilities, that I had nailed it -- but on playback it all sounded off. Not just a few notes, as a clumsy percussionist (me) might expect... the whole thing.

At that point I decided that maybe I didn't have "full hardware" compensation (it was added in the subsequent, most recent version of my DAW) and peformed a loopback ping which revealed my overdubs were being layed in 356 samples "late" -- behind previously recorded tracks. I nudged the track toward the beginning of the song by that many samples and then and only then did it sound like the part I'd recorded.

Admittedly, I've been analyzing very short delays for over a decade as I record, so I might be a trifle more attuned to these variations but I am hardly a rhythm robot... I think people may not KNOW where the problem is coming from, even that it's a timing issue -- but I think these "minor" timing misalignments and latencies really do contribute to a lack of precision in today's DAWs unless it's compensated for at the track alignment level (as Cubase, Tracktion, and now Sonar do; I think Logic might have such a "track alignment" compensation but I keep getting conflicting answers from Logic users.)


Anyhow, it's an interesting topic and it highlights some interesting facets of sound.

Maybe I'm just especially aware of it, being a subscriber to my local symphony orchestra and seeing seven completely unamplified orchestral concerts a year.

As many of you probably know or understand, the time it takes for the sound to go from front to back or side to side of a typical symphony layout can be as much as 40 or 50 ms or even more.

And it is for that reason that a large orchestra needs a conductor standing on a podium waving a visual rhythmic cue in the air. (Same thing for a drum major and marching band. Of course, there, the spread is probably far greater. I remember discussing this VERY issue in a class at my old school taught by a former HS marching band director. It suddenly made quite clear why so many marching bands get so out of time from front to back. If they don't watch the drum major's baton, they'll tend to play to what they hear, instead. And THAT can be a problem in a big band in a long, narrow formation.)

brentnowlan 7th February 2007 03:56 AM

Thanks for the responses. So to elaborate abit, right now I use the headphone output on the front of the 002 rack for the headphone mix, but the problem is that the headphone output jack is a mirror of what goes to the main out's, which makes what ever I hear they hear.
Ex. for doing bed tracks to a click track i have to blare the click track so they can hear enough of it in the foldback but i dont want to have to hear it in the control room. Thats why I want to use output 7-8.

Im not trying to add reverb while tracking. I'll try the thing Benmrx suggested about the buffer size and hope it works. kfhkh
I welcome more suggestion though

Will 7th February 2007 02:10 PM

I'm not familiar with the NZL in PT, but I guess you could make all of your tracks output to both outs 1+2 (for your monitoring) and 7+8 for the foldback by holding command when you select the outputs - I think this is the key, there is definatly a way of adding additional outputs to a track.

Then you could send the click only to 7+8 for them and have it as loud as they want. Only problem is that you won't hear it at all, unless you duplicate the click track and have your level on the duplicate fader coming out 1+2.

Hope this makes sense.

Harley-OIART 7th February 2007 02:21 PM

This thread will help with specific times at least.

https://www.gearslutz.com/board/music-computers/108056-digi002-latency-measurments.html

As for your question, I'm not sure i'm reading it correctly, but why don't you just monitor the foldback 7-8 along with the musicians?? Or if you want to make your own mix do that. Even behringer, my arch nemesis, makes a decent quality headphone distro with multiple input sources.

Anyways not sure If I read your question right.