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Lawrence wroteJeff told you they were already aware of it before you made the ticket. You didn't really expect to hear... "We'll change that right away because you new guys are way more important that the long time users who've already brought it to our attention six months ago!" ... did you? :)


:twisted: :mrgreen:

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by kenmatson on Wed Dec 06, 2017 8:25 am
kenmatson wroteI've got to do a little more testing in the morning and I'm not so darn tired! For now, a couple answers for you.

0) why wouldn't I? It's so that they can get low latency monitoring for what they're playing right? May not matter a lot for some of the instrumentalist, but it can mess a drummer or vocalist up pretty good to have much latency.

2) this is the one I got to check on to see exactly how it's behaving .... withholding further comment until then.


3) RME Firface 800. 64 samples getting me 11ms or so ... will experiment to see what I can get it down to. I know for vocalists, much over 5 or 6 milliseconds can start to feel funny.

thanks for bearing with my frustrations on this


Ok - I was wrong about #2 - I CAN still hear them live thanks to the standard old input monitoring ... I thought about this after I posted it while driving home last night, but wanted to verify this morning. So - my workflow can be to use Blue Z and then switch it off on mains for playback or punch. Still would like to see the bu ... issu.... uh .... "gap" fixed! LOL! But I think I can work this way in the mean time.

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by kenmatson on Wed Dec 06, 2017 9:03 am
jpettit wrote3) What interface are you using, most modern interfaces can get 10 ms or lower.



OK - playing around - going 96/24 - 128 samples - got it down to 6ms round trip - std monitoring - I think I can live with that with no problem ....

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by Lokeyfly on Wed Dec 06, 2017 9:08 am
Squashed that "bug".

Not sure you'd be seeing reduced latency at 96k, compared with a lower resolution.
But maybe we're back into the gap arena.

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by kenmatson on Wed Dec 06, 2017 9:29 am
Lokeyfly wroteSquashed that "bug".


Finding an acceptable workaround does not negate the bu...... "gap" LOL! It just makes it less painful ... like Novocain before drilling your tooth :mrgreen: :+1

Anyway - glad to having something I can work with - so on to my next issue - when I find one! THANKS AGAIN ALL for the help and patience as I get this working in my day-to-day workflow!
Last edited by kenmatson on Wed Dec 06, 2017 9:32 am, edited 1 time in total.

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by kenmatson on Wed Dec 06, 2017 9:32 am
Lokeyfly wroteSquashed that "bug".

Not sure you'd be seeing reduced latency at 96k, compared with a lower resolution.
But maybe we're back into the gap arena.


nope - latency goes down significantly at 96 vs 48 or 44 believe it or not. I've seen this before, and read some technical explanation of it at some point - anyway - works for me - 6ms is fine.

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by jpettit on Wed Dec 06, 2017 10:10 am
It is mathematical. A buffer is processed faster at 96,000 times per second than at 44,100 times a second. I will mark this as solved for the time being.

PS: Although he has heard it before from ProTools users, I have sent a PM to the product manager about functionality required in a multi-room studio.

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by mikemanthei on Wed Dec 06, 2017 6:19 pm
jpettit wroteI have sent a PM to the product manager about functionality required in a multi-room studio.


Thanks... I'm sure there are a few of us (im)patiently waiting.

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by kenmatson on Wed Dec 06, 2017 7:14 pm
jpettit wrotePS: Although he has heard it before from ProTools users, I have sent a PM to the product manager about functionality required in a multi-room studio.


Thanks!

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by Lokeyfly on Thu Dec 07, 2017 6:29 am
Buffer undersrood, Ken.
My point is its not much of a difference when recording even 96 k that will then go through plugins at some other resolution/frequency adding yet more latency (even during playback (switching) which you sounded pretty up in arms about. Or somebody's arms.

kenmatson wrote: ......at least for me - it's a show stopper for using LLM from S1. It's not just punching in - it's playback - e.g. Band records a take of a song - first thing they want to do is hear it back. I should be able to go back to song start marker, hit play, and they should hear it. instead - gotta turn off Z monitoring - great - easy - it plays ... but WAIT! - now I can't hear them - saying things like - "hey can you play that chorus again" ... or - "that's good enough, on to the next one" .... or ... "whoa - that sucks - trash that - we're gonna do another take" ..... instead - they fervently wave their hands back and forth and hope I see them through the glass! UPDATE! I WAS WRONG ABOUT NOT HEARING THEM WITH THE Z SWITCHED OFF - I NOTED THIS IN A NEWER POST- BUT THOUGHT I'D UPDATE HERE TOO!

Will try the high latency monitoring - but p'bly will just go back to my old method of using my interface mixer software - at least that works .. was really liking the idea of the integration and controlling all in S1 until this issue arose .... sigh


Nice flair for the dramatic.
Gotta turn off Z monitoring? Well yeah. Squeeze that in during musicians fervently waving hands. Add some effects on busses for good measure. That could be one switch.

The FR is in. You helped bolster that , so just be prepared to switch Z modes, or buy a quantum interface, or equivalent.

Yes, for vocals, you'll want to stay at or below 5 ms. If you feel you need to record at 96k, you'll still have additional latency to deal with when addimg real time effects. Typically, conv reverbs, add more latency than other plugings, along with a bigger hit on your Processor and memory (which is why I bring up investing in a faster interface).

It is in fact a "work around". Nicely adjusted!

I'll vote, as I'd be up for a little increased defining of playback/monitor/record switching.

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by kenmatson on Thu Dec 07, 2017 7:13 am
All I know is I can see it change every time I change the sample rate, and the lowest one is 96 with 128 sample size. I get about 6 milliseconds RT.. And I'm talking about non-z monitoring by the way. Oddly enough, I haven't seen any of my effects really change that much, even the convolution Reverb, but it's on a send.

As far as the waving the hands around fervently statement, I discovered that I was wrong about that, and corrected it in the original post and in another post. With non Z monitoring, I can hear playback, and I can hear the performers live, so with 6 milliseconds, I think I'm okay.

As far as the thread title, a moderator changed that, but I'm okay with it as it seems to make sense now.

As for being up in arms or dramatic, apologies, I don't mean to come across that way. It's just that I make a living with this stuff everyday, so if I'm in the middle of something and it's dragging me down, it can get me frustrated a little bit!

In any case, I seem to be okay at the moment with workflow. Hopefully I won't run into anything else. So far so good.

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by Lokeyfly on Thu Dec 07, 2017 11:50 am
Cool. No matter, you're trying to stay ahead of the demands, or just being ready. A good thing.

Maybe I missed some dialogue as well, but it sure sounds like you could use a Thunderbolt rated audio interface. You'd have the least latency, and ease of use.

Anyway, I'll still vote. Just away from that stuff right now.
Leave your link if you can for your FR. This way, others can vote as well.

I think you got some great help on the subject, from Jpettit. :thumbup:

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