Old 05-04-2017, 09:34 PM   #1
Crimson
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Default How far can you go pushing the latency wall?

Hi,

Been having fun jamming heavyguitar on a Drum Machine Vst and put the latency quite low on my Scartlett 2i4.

Running at 88200hz

The track is going through various plugins:
Distortion
Amp Head Emu
Cab Emu
EQ
Delay

The latency is quite short and doesn't annoy me so much and the sound is just perfect. Not crackels nothing.

THEN:
I record a loop on that track,
Create another track with quite the same plugins (different Cab lets say)

When I'm monitoring that track to play along with the drum and the other guitar, everything seems perfect for 5 to 10 seconds then the audio start crackling and crhhrrrrrrrrr rhrhhhrhrhr. If I stop-play, it's goes alright then starts going weird again.

What is so hard? I How can I play for hours without any problems on a track with live vsts along with a drum and then I get trouble when I add another guitar to play along the with first one?

I already met that kind of trouble on a 54 track'ish project and thought that I've been pushing a bit much.
I made through the end of the project with MAX latency and parallel monitoring but still, recording for more than 15 seconds at a time was impossible.

But encountering the same kind of behavior with 11 plugins (real toll) just doesn't scale in my opinion, there's surely something wrong. Anything sounds familiar? Please help

By the way,4.7ghz 8 core Cpu, 8gb of ram, SSD drive.


Thanks
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Old 05-04-2017, 10:27 PM   #2
Pickasso
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What is the ampsim you're running?
They're not all efficient!
Try a low quality setting if it has one.

Bypass any 'look-ahead' vsts like compressors while monitoring.

Try a stereo setup thru one ampsim instance while monitoring, see if that runs smoother.
Make a folder which has the fx on it. Add 2 tracks to folder to feed into it, pan hard l/R. Add something like js:dual paning device after the ampsim if you don't want a hard pan result.

Some good ampsims I like for low latency are Kuassa Creme, EZ mix, or Legion.

What buffer setting can you play at? 64 samples?
You could try 44.1 see if that helps too.
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Old 05-04-2017, 11:30 PM   #3
Judders
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When going for low latency, freeze all your tracks to free up processing power.
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Old 05-05-2017, 06:30 PM   #4
Crimson
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Using Legion on one, le cto on the other...

At the moment, It's doing it,
After I played for more than Half an our with abolutely no problem (one track leading while playing on the rythm guitar and drum. Changed nothing on the project since yesterday

was using 4 ms latency and 1024 spls

What I just don't get is WHY everything works fine then It starts being a nightmare wihout changes in parameters.

I know may be the block size is too big for the latency I'm using but it worked, then.... it stopped working no parameters changed.

Yeah 44.1 I guess could be a workaround but I got the ''impression'' that the sound is way clearer at 48k and over and I actually feel I can sculpt the sound more precisely when mastering, is it just me?

1-I can parrallel monitor if It could help? Is there a way to put the ''MAN! BUFFER 10 SECONDS AND NEVER CRACK'' setting somewhere?

2- If I'm sending to a REAmote over Gigalan, could it really help with that kind of issue?

I've been using StudioOne so much in the past that I really feel constrained when freezing a track. Studio One doesn't even have that feature. Don't get me wrong, I tried them all and I simply have faith in Reaper. I want this relationship to be long term and final.

Does anyone use a USB interface like mine and is able to get through with a 64+ track project without a Technical headache? What should I buy to have similar input quality with full bandwith and not having to monkey swing through turnarounds?

Thanks for the answers you guys are incredible

Chuck
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Old 05-06-2017, 02:32 AM   #5
Judders
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Quote:
Originally Posted by Crimson View Post
Does anyone use a USB interface like mine and is able to get through with a 64+ track project without a Technical headache? What should I buy to have similar input quality with full bandwith and not having to monkey swing through turnarounds?
Bandwidth doesn't work like that. The audio interface only needs to pass whatever audio inputs are coming in, plus the master output channels (which for most people will be 2 - stereo left and right).

Higher bandwidth connections allow more simultaneous inputs, but won't necessarily reduce latency to a great degree.

Quote:
Originally Posted by Crimson View Post
What I just don't get is WHY everything works fine then It starts being a nightmare wihout changes in parameters.
Could be OS background processes, or other apps.
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Old 05-06-2017, 08:03 AM   #6
Crimson
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Quote:
Originally Posted by Judders View Post
Could be OS background processes, or other apps.
Well well well.

Well

I'll investigate the background process hyppothesis, I guess I'm a fool to think that ''Computers nowadays are able to handle all that at once''.

I've been doing this in the past for my old Quad core and Vista Win 7.
Why audio workstations are more fragile to work with than Video processing software, or even game ws like Unity?

So could it be a nice idea to follow this procedure?:
https://support.focusrite.com/hc/en-...-on-Windows-10

I'll try this tomorrow

Thanks for your help, I'll bounce back if this is a life changer

Chuck
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Old 05-06-2017, 08:57 AM   #7
Pickasso
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Quote:
Originally Posted by Crimson View Post
Using Legion on one, le cto on the other...

was using 4 ms latency and 1024 spls

What I just don't get is WHY everything works fine then It starts being a nightmare wihout changes in parameters.
WOW! At 1024 spls, it should be unplayable (way too much delay), and you should be running totally glitch free @ 1024. I don't like more than 128 spls when record monitoring, 64 is more ideal.

There is a good possibility you have a buggy plugin if this happens at 1024 samples buffer. 2 questions:

Does this issue begin when there is no active playback?

Can you post your Reaper project (with no audio tracks), so someone can test your setup?
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Old 05-06-2017, 09:54 AM   #8
Crimson
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Quote:
Originally Posted by Pickasso View Post
WOW! At 1024 spls, it should be unplayable (way too much delay), and you should be running totally glitch free @ 1024. I don't like more than 128 spls when record monitoring, 64 is more ideal.

There is a good possibility you have a buggy plugin if this happens at 1024 samples buffer. 2 questions:

Does this issue begin when there is no active playback?

Can you post your Reaper project (with no audio tracks), so someone can test your setup?
Will do, tomorrow probably.

Surpisingly 1024 didn't seem to be an issue on my setup, I understand what you mean but everything seems very tight when I play it back and when I'm I'm monitoring, the reason it was still at 1024 was because of another project that was using 118 plugins and I was desperate.

Thanks Pablo
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Old 05-06-2017, 02:22 PM   #9
snooks
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Quote:
Originally Posted by Pickasso View Post
I don't like more than 128 spls when record monitoring, 64 is more ideal.
It's worth mentioning that saying that doesn't mean much by itself because the round trip latencies (RTLs) for interfaces are so different at the same buffer sizes.

eg an interface with a 64 sample buffer size can have roughly the same RTL as another interface at 256 samples. See the table in the first post in this thread for details...

https://www.gearslutz.com/board/musi...data-base.html
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Old 05-22-2017, 09:59 PM   #10
Crimson
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Hi

I started another project with lots of VSTs again, (always yeh)
I've been working the whole time at 88200 and had similar problems again, on and off.

When rendering the track, I realized that some ''Gravel's'' decimation effect didn't render as it should In 44100 so I modified the session render to 44100 in order to tune the VST correctly and yes I observed that everything was very smooth. When I switched back to 88200 it went stutter crazy again.

Can someone explain to me what is going wrong with 88200 that isn't in 44100?
Is there a possibility that PDC goes crazy when some plugins aren't particulary meant for 88khz when running with others that are All wheel drivers?

Or

Is it simply logical that it's ''Harder'' to sync all that stew on higher KHZ and you may say '' Don't even try running that much plugins at once beyond 48khz crazy boy nobody does!''

Thanks!

CC
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