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Will a hardware reverb cause latency as well?

using Logic X on a 2017 iMac full specs.

even a UAD Quad MKII thunderbolt with a lexicon reverb (or the seventh heaven) will still cause me latency when I put it on the main out.

Or is the fact that I put it on the main out the problem?
(I have EW Spaces as a bus plugin on all the sections, which doesn't cause significant latency.)
audiobuffer settings are medium and 512.
I need to check this later but I think I have latency compensation on with monitoring in Logic X.

Suggestions on either direction? hardware reverb with low latency possibility <---> or different way to set an "overall" reverb plugin on the main output without adding latency?
 
Last edited:

dgburns

summer of pickles and IPA beer
Your buffer setting will determine latency overall, soft or hardware verb. I use an outboard lexicon, but when I compare it to Valhalla or B2, with lower buffer settings, it’s all perfectly usable for tracking, imho. But it’s not like the UA cards with set latency on things like the new 480 which is around 48 ms no matter what.

BTW, not a fan of strapping a verb across a whole mix myself. Prefer sending from sources and using an aux as a return that goes to the master.
 

Dewdman42

Senior Member
using Logic X on a 2017 iMac full specs.

even a UAD Quad MKII thunderbolt with a lexicon reverb (or the seventh heaven) will still cause me latency when I put it on the main out.

Or is the fact that I put it on the main out the problem?
(I have EW Spaces as a bus plugin on all the sections, which doesn't cause significant latency.)
audiobuffer settings are medium and 512.
I need to check this later but I think I have latency compensation on with monitoring in Logic X.

Suggestions on either direction? hardware reverb with low latency possibility <---> or different way to set an "overall" reverb plugin on the main output without adding latency?
you are getting significant latency from your 512 buffer setting. Many reverb plugins add zero extra latency above and beyond that. If you try to use a hardware reverb, then you will get worse latency you are experiencing now because you have to go both out and back in through your audio device at a 512 buffer setting, the hardware reverb device itself will not be adding any appreciable latency.
 
Thanks for the mesages, which means that will not be the way forward.

I will see what different buffer settings will do again, but unfortunately the iMac begins to throttle the fan even when ot doesn’t do that much.......only a 80% load on the RAM, and half cpu.......

Thanks!
 

Dewdman42

Senior Member
Unfortunately latency is a hard reality of working with a computer DAW. You have to spend money if you want to get the latency down, and you can, but it will mean better audio devices probably more than anything, though a faster computer may also help reduce buffer size a bit, but IMHO it usually comes down to what audio hardware you are using as some have more efficient drivers then others and it depends on which bus its operating on, etc.

Something else to consider....

we really only care about latency when we are trying to actually record parts to tracks. Whether we are recording a midi part, or singing a vocal part, or recording some instrument.... if we hear latency while trying to do that, that's when it harms performance. Once you're in the mix stage it doesn't really matter what the latency is. Set the buffer bigger during that stage to give your cpu some breathing room.

So while recording tracks, first you have to decide how much latency you can tolerate. Some people can tolerate a lot, other people can barely tolerate any. I am of the latter category. There are several ways this manifests itself. One is in timing of what you hear. But to put that in perspective, if you were playing an electric instrument on stage and standing 10 feet away from the monitor, you would hear 10ms of latency and somehow you would probably adapt to that ok. But me personally I do notice a difference beyond about 10ms.

But further to that, some singers if they are using headphones will hear flanging in their ears if the monitoring sound they hear through the headphones mixes with the bones in their inner ear and the monitored signal has too much latency. That can be really annoying for some people too, not to mention the delay factor if its more than 10-15ms.

So one approach is that while recording tracks you minimize all latency, even if you have to temporarily use different FX while recording parts. Eliminate latent plugins for example. But even without any plugins, you still have the audio card latency, so what many people do, myself included, is try to monitor the thing you are recording WITHOUT going through the DAW mixer while you're recording it. You have to use some kind of external monitoring mixer and probably some external FX to use temporarily while you record the tracks. You run your vocal or instrument or whatever, through that external mixer with essentially zero latency. Its getting recorded into the DAW but you're not listening through the DAW, you're listening through the external mixer. After you record the track, you you listen to it through the DAW of course and use your plugins as you wish. That can help a lot in terms of some of the annoyances while recording.

If you spend money you can probably get your DAW down to RTL of around 5ms, realistically speaking. Much lower then that is mostly theoretical. In practical terms you're probably going to be doing well if you can record tracks through your DAW with 5ms Round Trip Latency (actually measured, not what the DAW reports which is not always accurate).

I spent an extra $999 to get the latency of my X32 mixer down to 3.8ms RTL at 64 sample buffer, had to buy a Lynx PCI card to help with that. Over USB it was double or triple that and couldn't really work well at 64 sample buffer either. The really low latency solutions like Lynx, RME, MOTU and some others...they cost money...simple as that. And a faster computer allows you to drop the buffer size down a bit, but still depends a lot on how efficient the audio device driver is to do that. You will see reports from some devices that they can go down to 1.6ms or other really low numbers, but keep it in perspective, for the last 10 years latency has really not effectively lowered much, because those low reported numbers are extremely theoretical. For example MOTU has a device that reports 1.6ms latency, but that is running with a 16 sample buffer at 96k. You will need a very fast computer to do that and probably will still get drop outs in real world use with a 16 sample buffer. Also, using 96k sampling cuts the latency in half...so realistically, if you're at a 32 sample buffer at 48k, the latency of that device is most definitely not going to be 1.6ms.

The reality is that 5ms RTL is just about as low as anyone can expect to get out of their computer-based audio device in practical terms, IMHO and you have to spend some bucks in the thousands to get that. If you can live with 5ms while monitoring yourself through the DAW, then great. If not, then you want to use an external mixer and monitor through the hardware while tracking, using temporary fX rather than plugins.
 
Last edited:

chimuelo

Star Of Stage & Screen
I mix hardware Reverbs by taking AUX Sends and Returns in your software mixer out and back in to selected channels.
I use real time DSP App Mixers but I don’t see why RME Total Mix couldn’t do this.

But I don’t use ant Native FX, prefer outboard processing.
Like my Computers to stream samples and audio, hardware FX are for real time.
Pretty much makes latency a thing of the past.

Here’s my entire Hardware FX set up.
Strymon Mobius for everything related to Modulation FX, Strymon Timeline for Delays from Heaven and Hell, and Strymon Big Sky for any Reverb you’ve never heard of.
All MIDI Automated, zero latency, just great shit for any DAW but this is a live rig, zero latency isn’t necessary but it’s there if you want it.

77431540-3371-44EB-AB6E-EABF567D5151.jpeg A8C229B8-375A-476D-919D-1B32FC96DFF1.jpeg
 

JamieLang

Active Member
You need to use your UAD reverb inside CONSOLE....not Logic--you actually HAVE low latency hardware reverb unit. That's what an Apollo is...

ALL your cue should be controlled via Console and not Logic.
 

benatural

Active Member
Unfortunately latency is a hard reality of working with a computer DAW. You have to spend money if you want to get the latency down, and you can, but it will mean better audio devices probably more than anything, though a faster computer may also help reduce buffer size a bit, but IMHO it usually comes down to what audio hardware you are using as some have more efficient drivers then others and it depends on which bus its operating on, etc.

Something else to consider....

we really only care about latency when we are trying to actually record parts to tracks. Whether we are recording a midi part, or singing a vocal part, or recording some instrument.... if we hear latency while trying to do that, that's when it harms performance. Once you're in the mix stage it doesn't really matter what the latency is. Set the buffer bigger during that stage to give your cpu some breathing room.

So while recording tracks, first you have to decide how much latency you can tolerate. Some people can tolerate a lot, other people can barely tolerate any. I am of the latter category. There are several ways this manifests itself. One is in timing of what you hear. But to put that in perspective, if you were playing an electric instrument on stage and standing 10 feet away from the monitor, you would hear 10ms of latency and somehow you would probably adapt to that ok. But me personally I do notice a difference beyond about 10ms.

But further to that, some singers if they are using headphones will hear flanging in their ears if the monitoring sound they hear through the headphones mixes with the bones in their inner ear and the monitored signal has too much latency. That can be really annoying for some people too, not to mention the delay factor if its more than 10-15ms.

So one approach is that while recording tracks you minimize all latency, even if you have to temporarily use different FX while recording parts. Eliminate latent plugins for example. But even without any plugins, you still have the audio card latency, so what many people do, myself included, is try to monitor the thing you are recording WITHOUT going through the DAW mixer while you're recording it. You have to use some kind of external monitoring mixer and probably some external FX to use temporarily while you record the tracks. You run your vocal or instrument or whatever, through that external mixer with essentially zero latency. Its getting recorded into the DAW but you're not listening through the DAW, you're listening through the external mixer. After you record the track, you you listen to it through the DAW of course and use your plugins as you wish. That can help a lot in terms of some of the annoyances while recording.

If you spend money you can probably get your DAW down to RTL of around 5ms, realistically speaking. Much lower then that is mostly theoretical. In practical terms you're probably going to be doing well if you can record tracks through your DAW with 5ms Round Trip Latency (actually measured, not what the DAW reports which is not always accurate).

I spent an extra $999 to get the latency of my X32 mixer down to 3.8ms RTL at 64 sample buffer, had to buy a Lynx PCI card to help with that. Over USB it was double or triple that and couldn't really work well at 64 sample buffer either. The really low latency solutions like Lynx, RME, MOTU and some others...they cost money...simple as that. And a faster computer allows you to drop the buffer size down a bit, but still depends a lot on how efficient the audio device driver is to do that. You will see reports from some devices that they can go down to 1.6ms or other really low numbers, but keep it in perspective, for the last 10 years latency has really not effectively lowered much, because those low reported numbers are extremely theoretical. For example MOTU has a device that reports 1.6ms latency, but that is running with a 16 sample buffer at 96k. You will need a very fast computer to do that and probably will still get drop outs in real world use with a 16 sample buffer. Also, using 96k sampling cuts the latency in half...so realistically, if you're at a 32 sample buffer at 48k, the latency of that device is most definitely not going to be 1.6ms.

The reality is that 5ms RTL is just about as low as anyone can expect to get out of their computer-based audio device in practical terms, IMHO and you have to spend some bucks in the thousands to get that. If you can live with 5ms while monitoring yourself through the DAW, then great. If not, then you want to use an external mixer and monitor through the hardware while tracking, using temporary fX rather than plugins.
You mentioned needing to spend thousands. Curious about this, what gear would you need to buy?
 

AlexRuger

Senior Member
I mix hardware Reverbs by taking AUX Sends and Returns in your software mixer out and back in to selected channels.
I use real time DSP App Mixers but I don’t see why RME Total Mix couldn’t do this.

But I don’t use ant Native FX, prefer outboard processing.
Like my Computers to stream samples and audio, hardware FX are for real time.
Pretty much makes latency a thing of the past.

Here’s my entire Hardware FX set up.
Strymon Mobius for everything related to Modulation FX, Strymon Timeline for Delays from Heaven and Hell, and Strymon Big Sky for any Reverb you’ve never heard of.
All MIDI Automated, zero latency, just great shit for any DAW but this is a live rig, zero latency isn’t necessary but it’s there if you want it.

View attachment 20955 View attachment 20956
Dude, every time you post about your rig it cracks me up because I never see the same gear twice. I feel like you have every piece of gear and every CPU ever made.
 

chimuelo

Star Of Stage & Screen
Dude, every time you post about your rig it cracks me up because I never see the same gear twice. I feel like you have every piece of gear and every CPU ever made.
It’s a wall of racked PCs and hardware.
There’s a permanent recording rig, a rehearsal rig, then the live rig.
This is my air cooled summer fair/outdoor rig.
 
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