HUGE LATENCY ISSUE.

[DwayneW7]

Im very new to using ableton and am not familiar with all the terms yet so please excuse my audiolingo mistakes!!

I have 2 devices (one a gaming laptop, one is a gaming pc. Great specs on both) I run ableton on, both on live 11 suite and both are windows 11. The issue is, when I turn on ableton and plug in my devices (zoom v3 vocal processor, samson co1upro usb mic, shure sm58, IEM system) I turn plug in my in ears to the output monitoring of my co1upro (i use that as my IEM output, and the aux out as my master) I get like 6+ms delay which I feel is pretty bad because my voice sounds like its doubling when I hear the IEM out and my actual raw voice when I have one earpiece out. Ive been trying to figure out why the latency has been so garbage. Everything I try is not working, the buffer size is at 64, the driver error compensation doesn’t work, audio is at high quality and is at 88200 sample rate. Im not sure what else to do. My friend is telling me that maybe getting a better mixer would help but im not sure what a good MULTITRACK usb mixer I should invest in, also would prefer one that has those usb audio return buttons and is around like $200-500. I haven’t been able to find a good one yet but maybe that would fix my troubles? A better mixer? A better setup for my audio outs? Ive pretty much been jerry rigging it and maybe thats why my latency suck LOL. Please grant me the knowledge of the audiotech gods and tell me something else I can do to fix this frickn latency. Ill probably invest in a good multitrack usb mixer soon.

Thank you for ur time

[yur2die4]

Is this for music? Are you recording where you need to synchronize to other content in the project? Or is your voice going to be the primary factor in pacing?

If you’re doing music, then a lot of the things you’re looking at make sense (though the 88200 sample rate seems weird, maybe there a new thing?)

If you’re doing spoken word or dialog that isn’t directly supposed to sync to anything else, then you might be able to take advantage of ‘direct monitoring’ rather than trying to listen to the output of the software. You can listen to the output afterwards.

Additionally, if you have certain plugins, they can add extra latency to the set, so the 6ms would be for your computer with interface, but some reverbs, vocal processors, mastering plugins could easily change that to 26ms.

[DwayneW7]

This is for music . I made sure my plugins dont add extra latency, and made sure that everything is working at the lowest possible latency. Maybe a better usb mixer will help with the latency? Cause what im using right now is a usb mic and a normal vocal processor. After everything else I do, i still cant have it have a below 4ms delay.

[grrrz]

6ms latency round trip is pretty tight; upping sample rate and lowering buffer will make it drop (if your computer can manage) but you need a high quality sound interface to manage this without errors. 6 ms should be somewhat usable; but that's why a lot of sound interfaces have direct monitoring possibilities so you can ear yourself without any latency. don't you have a way to do that? not sure I understand what you're doing.
digital hardware processors can also produce a bit of latency (usually very low) so check every part of the chain.

[Tarekith]

6 ms is VERY useable, it’s the same latency you get sitting 6 ft from your speakers. Roughly.

[chris harbin]

Am I daft, or is this possibly a monitoring issue? 6ms is really low (especially if that is round trip)

[Calagan]

6ms is very good indeed and you don't need to worry about that. Your problem is more a sound engineering issue.

If you don't want to hear doubling on your voice, you need NO LATENCY at all and use the direct monitoring of your audio interface.

Actually, even 1ms delay is causing comb filtering if the processed signal is mixed with the natural source, and this comb-filtering/doubling effect is quite noticeable on vocals : the issue with vocals is that you hear your own voice from the inside, and it sounds strange when it's mixed with a copy of it with a little delay. Even using an analog console, to hear its own voice doubled in the headphones or (even worst) in a speaker is strange, and there's no delay on analog (just the tiny delay the sound taks to go from the source to your ears)...

So here are some fixes :
- use direct monitoring from your interface so you can hear your dry vocal with no latency at all
- if you need to process your voice, you can use without any issue reverb, delay, chorus or other time processors, because their purpose is anyway to mess with time and they can be added to the unprocessed signal without any issue
- if you need to use EQ, compression or other similar FX, you'll need to deal with the latency. One solution is to get rid of any direct monitoring and dry signal, and monitor your processed vocals sufficiently loud so you don't hear anymore the "inside voice". My experience is that it's working ok with 6ms of round trip latency (or even more, until 10ms I guess), especially when the vocals are quite heavily processed (with saturation or colorful processing, so it sounds different from the "inside" voice). Be careful not to monitor to loud and damage your ears !!!
- Also, don't hesitate in experimenting with different latency values. Sometimes, and depending on your voice and what kind of processing you need, 8ms works much better than 4ms for exemple, because the comb-filtering is affecting different frequencies.

In order to understand what comb-filtering is doing with your audio, you can experiment with a very simple test :
- on track A, use any source you like, vocals for exemple
- copy this source on track B, and delay it from 1ms to 50ms (using a delay plugin with zero feedback or the track delay feature of Ableton Live)
- play the two sources together, and enjoy the results...

EDIT : re-reading you post, you may also have issues with your IEM system. Is it wired or wireless ?
And also, is the USB microphone adding its own latency ? I'm not familiar with these devices...
The strange thing in your set up is it looks like you're using more than one USB device that is supposed to act as an audio interface. Do you have the same issue if you plug your microphone (with an XLR cable) in your interface and then monitor it with the headphone out of the interface ? Or, better explained, if you get rid of the USB microphone and the IEM system ?
Because it's not clear what interface you are using in order to communicate with Live for the in/out... In your case I would use the Zoom as audio interface, plug a microphone in it with a XLR cable and use the zoom outputs as monitoring, but even after re-reading your post I'm confused by your signal chain...

[NataieMoon]

So, this is all consuming me lately! For years, I've recorded with a Neumann and haven't had any issues with latency. I record at 32 ms and I have the monitor on because I need to hear any small noises. I also like to have some reverb on while I record. I recently bought a Shure Mic and now it's latency city even though I'm using the exact same settings as I did for my Neumann. I'm wondering if it's a microphone thing. Let me know if you fixed the issue.