Tracking in LogicPro7 with LA2A

[xist2005]

Hi, did anyone try the new Live (in UAD's driver 4.3) option to track with an LA2A in Logic?
Could you give me your opinions, and tips for setting everything in the right way (for istance: how to set up Logic's PDC in these cases?)
Thanks.

 

[HKC]

I haven´t tried it with UAD1 but I would set Logic to \"tracks and inst\" and the lowest possible buffer range. This works on everything else but why are you so interested in tracking with LA2A. You do know that it won´t work like an outboard comp/lim would. You can´t get hotter levels with a plugin only outboard stuff before the converter can do that.

 

[xist2005]

So it's useless to track a voice that way?

 

[thewitt]

I haven't tried this 'live' option yet (I'm always slow to upgrade), but as for the second question:

Yes, you can use it like an outboard comp, just insert it on an input.

I believe HKC is speaking of inserting on a channel strip, which is non-destructive.

 

[arimaka]

What HKC is saying is that no matter what you do (even putting the plug on the input) you will always be AFTER the A/D converter, so the advantage of an outboard compressor (to track vocals a little hotter, while controlling their dynamic range) is never there. You can always track vocals with the effect but it would be identical to putting the effect on the track. It will be after the A/D, jut maybe printed to the audio file that's it.

 

[demonsp]

The benefits of getting maximum level to DAW (by compressing, EQing) is attained BEFORE hitting the AD stage.

As arimaka points out.

The idea is to channel strip your sound with outboard equipment, allowing maximum bit usage. (ie. use all 24 bits if that's the format chosen)

Thing is, when recording at 24 bits, you have so much headroom available
(according to my research) that you'd have to get extremely poor levels to \"tape\" (DAW) before your sound suffers.

There is benefit though to get maximum levels. Especially when DSP processing.

 

[xist2005]

Uhm, I did not get the last post, sorry.
So what's the point, in your opinion, on having this live option?

 

[HKC]

If you use an amp simulator like Nigel your whole performance depends on that you have the right (ish) sound while recording. On such occasions you will need a low latency (below 256, I´d say the least) to get the feel right. This is where this new live mode comes in handy. Before 4.3 it wasn't really possible to use Nigel as a recording tool but now you can if your computer is up to it. If not then just use Logics native guitar amp and change it to Nigel afterwards if that's more to your liking.
I could also picture a situation where you have recorded something, treated it with ie a Fairchild and then you realise that you need to add or redo a part. With the livemode you don't have to bypass the Fairchild to get a decent latency but of course the bypassing is actually faster than changing to livemode if it's not already in it.
In Powercore each plugin has a \"livemode\" button which is a good idea but the fact is that if you have some serious plugins and virtual instruments running at the same time you will pretty quickly run in to problems with the much dreaded pops and crackles.
The solution is still to use dsp based plugins for post recording only unless you are at the beginning of a project and your processor still has plenty of power available. This is not a big issue to me because Logic pretty much covers all territories well so I just use them while recording and then add the dsp hungry plugins when I´m done.

 

[JuergenW]

Thing is, when recording at 24 bits, you have so much headroom available
(according to my research) that you'd have to get extremely poor levels to "tape" (DAW) before your sound suffers.

Absolutly! Using (analog) limiting and compression while tracking
is only necessary when the dynamic range of the media is lower
than the dynamic range of the recorded source. As for 16 Bit it is 96 dBFs
dynamic range one might want to get hotter average levels by
analog limiting. As each 6 dBFs yields one Bit resolution (16x6=96)
in 24 Bit you have 24x6=144 dB dynamics.

Nearly no A/D converter offers better dynamic ranges than let's say
122 dB. Which means one has usualy 24 dB headroom before loosing
effective dynamic range of the analog source - except in live and broad-
casting situtations there is no technical reason to track with
analog compression/limiting, only estetic ones...

Tracking with digital dynamics after conversion brings no technical
advantage, except saving resources or in special cases for monitoring.

 

[pogo]

If you really want to record with plug-ins (or bounce with external sources), create and use Input Objects in your Environment. Page 297 of the manual explains the process.

 

[F5D]

In Powercore each plugin has a "livemode" button which is a good idea but the fact is that if you have some serious plugins and virtual instruments running at the same time you will pretty quickly run in to problems with the much dreaded pops and crackles.

I haven't tried the uad1 logic live mode yet (will try soon) but does it really mean the same live mode as powercore live mode which uses tons of cpu to enable a kind of instant communication with the dsp card? What I understood from the UA report is that the live mode reduces the huge latency (2048 samples or 1024?) which uad1 plugins normally introduce in logic and when live mode is enabled, uad1 plugins work more like powercore plugins and the introduced latency changes when you change the audio interface's core audio latency? I guess this is the case if native cpu usage doesn't jump 10-50% per plugin when using uad1 live mode. Of course it will use more native cpu too when decreasing latency, just like powercore plugins in normal mode, if it works the same way.

Also have you noticed much difference between the live mode and normal mode when running big projects? Does the live mode use more native cpu when loading uad1 plugins? Can your projects load plugins to 2 cpu's at the same time when using the live mode as in normal mode? At first I understood from the UA note that all the plugins will run on only 1 native cpu but later I was thinking if it only means per audio track? So, if track 1 has eq + comp + delay + reverb (for example 1 uad1 plugin and the rest native), they all use only 1 cpu but track 2's plugins could already use cpu 2 ?

Anyway, I will test the live mode soon when I'll get back to my studio.