Anyone do serious tracking w/software monitoring and plugins?

[klong]

I've been in the Universal Audio ecosystem for a while, and it's great, but I'm tired of managing presets for UAD interface/DSP and Reaper, and I'm obviously not switching DAWs to their LUNA system, as neat of an idea as it is.

I like to record vocals with compression and reverb etc, and guitar with good amp sims and whatnot.

But I have never found software monitoring to 100% satisfactory. I guess if you are careful about plugin selection and freeze tracks you aren't actively recording, you can get away with a 32 or 64 buffer on a good computer/interface, but when you get cracks/pops it's pretty darn annoying.

The Presonus Quantum 2626 interface looks interesting, under 1 millesecond of latency supposedly. The interface has no DSP or even hardware monitoring, it's supposed to be just lightning speed into your DAW.

That won't resolve the issue of running non-real-time-capable plugins or crackles and pops though.

Hmm.

[Wallaby]

Have you tried an RME interface ?

[serr]

You can be matter of fact with the setup.

Start with a max latency you could put up with monitoring and recording with. That will likely be in the 3 to 11ms range for most people. ie. Where you no longer perceive a lag.

48k sample rate is usually the sweet spot for latency vs CPU hit. (96k crosses the line for CPU hit even though half the latency per sample.)

You dial down the block size until you hit your target required latency. Some plugins have an internal latency. So any plugin with a higher internal latency than the system block size can't be run live. Many plugins are zero additional latency though. After that, the CPU time you have left is what it is.

That leads to shopping for a faster interface first for live stuff. That lets you land on a higher block size to hit your target latency requirement. That leads to bigger plugins live and more available CPU.

The target latency...
Some people are cool with 9 - 11ms. Some claim 3ms or they can hear it. Not to start up that argument here. Just whatever your target is, this setup can be pretty matter of fact.

One of the newer-ish thunderbolt connecting interfaces will usually have the lowest baseline latency. (ie lower than another model with the same computer with the same block size set in the DAW)

If you can hit your target at 128 sample block size, you'll be golden for just about anything. If you have to dial down to 64 or even 32 samples to make that latency, the system night be a struggle.

[future fields]

I monitor everything through REAPER. Using amp sims and what not. Never had an issue. Just pick the right plugins. Don't put a latency inducing look-ahead limiter on your tracking template.

You can use latency as an excuse but at the end of the day, it's no different than standing 10-20 feet from an amp. The tradeoff being that you have an entire studio at the clicks of a mouse. I'll take that tradeoff.

[klong]

Originally Posted by Wallaby

Have you tried an RME interface ?

I've definitely considered getting one and have read nothing but good things for years. The presonus quantum 2626 and 4848 claim to be the lowest latency of all.

I really love the build quality and DSP plugins of my Apollo Twin X, and have had other UA interfaces in the past. I've just really wanted to switch to a DAW-only workflow with no other Apps to muck with.

[Stephane Morisse]

I've had major problems involving Presonus products in the past, not even mentioning the fact that their drivers weren't maintained when a product vanished from the line.

[norbury brook]

Originally Posted by serr

You can be matter of fact with the setup.

Start with a max latency you could put up with monitoring and recording with. That will likely be in the 3 to 11ms range for most people. ie. Where you no longer perceive a lag.

48k sample rate is usually the sweet spot for latency vs CPU hit. (96k crosses the line for CPU hit even though half the latency per sample.)

You dial down the block size until you hit your target required latency. Some plugins have an internal latency. So any plugin with a higher internal latency than the system block size can't be run live. Many plugins are zero additional latency though. After that, the CPU time you have left is what it is.

That leads to shopping for a faster interface first for live stuff. That lets you land on a higher block size to hit your target latency requirement. That leads to bigger plugins live and more available CPU.

The target latency...
Some people are cool with 9 - 11ms. Some claim 3ms or they can hear it. Not to start up that argument here. Just whatever your target is, this setup can be pretty matter of fact.

One of the newer-ish thunderbolt connecting interfaces will usually have the lowest baseline latency. (ie lower than another model with the same computer with the same block size set in the DAW)

If you can hit your target at 128 sample block size, you'll be golden for just about anything. If you have to dial down to 64 or even 32 samples to make that latency, the system night be a struggle.

Good advice.

Regarding latency and what's comfortable for people.

I've found that playing a VI from a keyboard is a lot more tolerant of latency than singing with headphones.

I can play a Vi piano with 128 samples on my AXR4 no problem, however if I'm monitoring through Reaper and singing with Headphones I have to work at 32 samples or I can feel the latency.

Same with Guitars sims I need 32 samples or it doesn't feel like playing through an amp or my Kemper.


I'd recommend RME PCIe HDSP AIO pro plus something like a ferofish AD/DA 16 channel convertor.

Lowest latency drivers in the business and the ferofish is a great sounding 16 AD/DA for a great price. The 2 together wil cost a lot less than an UA Apollo and will sound better,have better drivers(latency) and also the ferofish can be use with any ADAT interface for ever more. So you could use it with the rme AIO and also with any portable interface on a laptop too.



M

[Hermetech Mastering]

No probs here recording electric guitar with low latency. Tone King Imperial plugin, RME HDSPe AES at 96kHz, with a 64 sample buffer (it won't go any lower at 96k). Never any dropouts/glitches, feels like playing through an amp.

[Glennbo]

Originally Posted by serr

Some plugins have an internal latency. So any plugin with a higher internal latency than the system block size can't be run live.

In addition to the block size there is also the number of periods, which you can change in Linux, but I believe are still fixed in Windows.

REAPER in Windows (last time I ran it) was fixed at two periods on the output, and one period on the input which makes the input single buffered and the output double buffered, so a 64 sample block size with two periods will result in a 128 sample window for REAPER to work with.

Many plugins are zero additional latency though. After that, the CPU time you have left is what it is.

Most of REAPER's native plugins are zero latency, so I always use them first if possible. I tested my machine with a loopback cable and it is dead on at 8ms round trip latency. I record all my tracks monitoring live through REAPER usually with lots of FX on the track, but always using very low latency plugins.

[norbury brook]

there's a handy Action in the actions list, bypass all plugins over 64 sample latency.

Put it on a toolbar an use it when tracking


M

[Glennbo]

Originally Posted by norbury brook

there's a handy Action in the actions list, bypass all plugins over 64 sample latency.

Put it on a toolbar an use it when tracking


M

Can it also detect plugin chains that exceed 64 samples?

That would be cool, but either way it sounds like a useful action.

[EirikG]

Originally Posted by klong

I like to record vocals with compression and reverb etc, and guitar with good amp sims and whatnot.
But I have never found software monitoring to 100% satisfactory. I guess if you are careful about plugin selection and freeze tracks you aren't actively recording, you can get away with a 32 or 64 buffer on a good computer/interface, but when you get cracks/pops it's pretty darn annoying.
Hmm.

In my experience the driver & interface & choice of communication-standard (iow not usb1/2 ) is the most important part, if computer otherwise has a decently fast infrastructure.

I've monitored through software in 48khz with both 64 &128 samples since 2002. (mostly 128 samples - still do ) All the time using Lynx interfaces. Now Aurora(N) on thunderbolt (2).
Sometimes it is challenging if that last addon-overdub happens when your project is nearing mix (has lots of heavy plugins running) - but in reality it just needs me to freeze some stuff to do it, then unfreeze after. Obviously avoid latent & heavy plugins in record. Plenty of great alternatives for the recording task!

[EirikG]

Originally Posted by EirikG

I've monitored through software in 48khz with both 64 &128 samples since 2002.

And yeah, it's been "serious" :-) - professionally recording live bands in my studio(s) - mostly 24 simultaneous tracks, but also 32 tracks through Auroras TB since I switched to Reaper. Probably 30-50 plugins running "live" & feeding monitoring & cans.

[bigtop]

I'm also in the UAD ecosystem with Apollo Twin MkII thunderbolt. I track vox through the UAD SSL 4000 channel strip console unison insert and reverb/delay as send/return on console aux 1/2. There is no audible latency during record which as you know is the whole point of console.

I save the UAD settings as presets and I save the console set as a preset so I'm up and running in seconds.

The only monitoring through REAPER is VSTi such as Omnisphere and kontakt and in that case the latency at 512/265/128 is not noticeable even during fast keyboard runs. This seems to be the case in most VSTi setups because;

MIDI travels light and travels fast
MIDI doesn't make any sound that "competes" with the VSTi sound (unlike trying to monitor vocals through DAW with reverb/delay).

Yes I know there's work arounds but they don't really work all that well.

In short I don't find it a chore to save/load UAD presets and the UA set up works well for me (I know Norbury disagrees!)

[norbury brook]

Originally Posted by EirikG

And yeah, it's been "serious" :-) - professionally recording live bands in my studio(s) - mostly 24 simultaneous tracks, but also 32 tracks through Auroras TB since I switched to Reaper. Probably 30-50 plugins running "live" & feeding monitoring & cans.

cool, this also brings up the point not every manufacturers drivers are the same when comparing buffer size.

Lynx,RME have very low actual latency at 128/64 samples. Some other manufacturers will have a real time latency higher than this even when choosing say a 32 sample buffer, so bear this in mind too.

The quality of the drivers, in combination with your computer specs will determine your low latency ability.

regarding UAD it's a fine ecosystem if you want to buy into it, and works well if you work 'their' way and use their console and plugins which are obviously using the DSP and are low (zero) latency.

routing out monitor mixes and tracking a live band this way would be a headache though compared to doing it all in reaper which Eric G is doing.

You'd be adding a whole layer of complexity , you may as well go the whole hog and use the Luna system DAW at that point.



M

[King Stupid]

Originally Posted by norbury brook

cool, this also brings up the point not every manufacturers drivers are the same when comparing buffer size.

Lynx,RME have very low actual latency at 128/64 samples. Some other manufacturers will have a real time latency higher than this even when choosing say a 32 sample buffer, so bear this in mind too.

The quality of the drivers, in combination with your computer specs will determine your low latency ability.

regarding UAD it's a fine ecosystem if you want to buy into it, and works well if you work 'their' way and use their console and plugins which are obviously using the DSP and are low (zero) latency.

routing out monitor mixes and tracking a live band this way would be a headache though compared to doing it all in reaper which Eric G is doing.

You'd be adding a whole layer of complexity , you may as well go the whole hog and use the Luna system DAW at that point.



M

From what I understand, every (the vast majority?) of interfaces have a "safety offset" buffer built in that can't be changed. My motu, for instance, is 16. I've heard that many have much higher safety offset buffers, some as high as 256 or even more. A safety offset of 128 would add ~5ms at 44khz on top of whatever your adjustable buffer settings are. Could be terrible for real-time recording situations.
When I learned this, I wondered why this "feature" is not listed in an interfaces specifications.

[norbury brook]

on windows and ASIO most good audio interface drivers from the likes of Lunx, RME etc have no safety buffer.

On OSX with core audio , especially with class compliant drivers there's often a safety buffer.

My Antelope has a check box in the ASIO settings where you can choose the safety buffer or not. it only adds 1 or 2 MS to the output , can't remember what but it's no where like 5.


Same with my AXR4 interface, there are 3 settings per buffer size:

1: lowest latency=no safety buffer
2; Standard= small safety
3: safe= higher safety



So you have a choice. Linux with ALSA lets you choose .


I set up 3 systems last year with the same UR-22c interface.

windows 11
OSX Big Sur
Linux manjaro

the linux and window were on the same i7 machine and the OSX was on an i7 macbook pro of the same era, the mac book had twice the ram.

For the same buffer setting the macbook had the worst (highest) real time latency, followed the windows 11 machine in the middle and finally the linux was the lowest.




M

[King Stupid]

Originally Posted by norbury brook

on windows and ASIO most good audio interface drivers from the likes of Lunx, RME etc have no safety buffer.

On OSX with core audio , especially with class compliant drivers there's often a safety buffer.

My Antelope has a check box in the ASIO settings where you can choose the safety buffer or not. it only adds 1 or 2 MS to the output , can't remember what but it's no where like 5.


Same with my AXR4 interface, there are 3 settings per buffer size:

1: lowest latency=no safety buffer
2; Standard= small safety
3: safe= higher safety



So you have a choice. Linux with ALSA lets you choose .


I set up 3 systems last year with the same UR-22c interface.

windows 11
OSX Big Sur
Linux manjaro

the linux and window were on the same i7 machine and the OSX was on an i7 macbook pro of the same era, the mac book had twice the ram.

For the same buffer setting the macbook had the worst (highest) real time latency, followed the windows 11 machine in the middle and finally the linux was the lowest.




M

Very interesting. Thank you for that information.

I wish I had tons of money so I could buy "all the things" and test them. This stuff fascinates me.

I had a different experience with my motu, on MacOS via TB, it is .3ms lower latency than windows 10 via TB. My old 18i20 was also lower latency than on windows via USB, but I can't remember how much.

The motu software shows the safety offset, which is 16, on windows, and though it looks like you should be able to turn it off, you cannot. Perhaps in an update sometime. Though it's stable at 32/96, so not really an issue.

[brk303]

Originally Posted by klong

The presonus quantum 2626 and 4848 claim to be the lowest latency of all.

Lowest latency sounds good, but only if you can actually run decent amount of plugins at that latency, read this if you haven't already.

https://gearspace.com/board/music-co...data-base.html

[redddog]

I run a Presonus OG Quantum 2632. It runs @ 1.9ms and sounds amazing. The preamps are some of the best clean pres I've ever used.

Never as much as a hiccup from this set up.

Crappy Presonus development is true on the older stuff - the firepod was an absolute disaster - but the TB stuff is a true step forward for them. Truly pro kit.

Don't cross them off the list for past crimes.

[klong]

Originally Posted by bigtop

I'm also in the UAD ecosystem
....
In short I don't find it a chore to save/load UAD presets and the UA set up works well for me

This is basically where I keep arrriving back at after every attempt to ditch the Console/DSP workflow.

Nothing I have ever tried works as well as using UAD plugins in the Console and printing them to REAPER. I also use the AUX tracks so I can print a wet and dry version of whatever I'm recording into REAPER>

[norbury brook]

Originally Posted by klong

This is basically where I keep arrriving back at after every attempt to ditch the Console/DSP workflow.

Nothing I have ever tried works as well as using UAD plugins in the Console and printing them to REAPER. I also use the AUX tracks so I can print a wet and dry version of whatever I'm recording into REAPER>

problem is you wont know what it's like to do it all through software using a UA interface as their low latency drivers aren't very good. They want you to use the console and monitor that way so don't bother with good low latency.

i sent my Apollo back because of this and being told what I've just told you by UA tech support!!

Get a lynx/RME/presonus(TB) and you'll be able to work 100% inside your DAW only in reaper probably @64 sample buffer with those interfaces.

Also remember 64 sample buffer with lynx or RME is what 128/256 is on lesser interfaces in regards to actual latency.



M

[Stephane Morisse]

RME have pretty solid drivers. I use a USB2 Fireface UC and have no issue at all. Of course make sure "use audio driver reported latency" is checked AND add more finesse using Reainsert Ping to add a relevant sample value.