Logic Pro 9 Who is using a 32 sample buffer rather than direct monitoring?

[Ru_C]

I'm interested to hear what others are doing.

I'm trying to reproduce the feel of my TDM rig, with a native Logic setup.

So far i've been more than impressed with the performance of Logic 9.1

I've tried it with 3 different interfaces, & all have performed fine (in my limited testing) with the buffer set to 32 samples.

Whilst I imagine this still gives a slightly bigger Latency than PT HD, it's certainly in the ballpark of usability.

Are Totalmix, Cuemix etc now redundant? Are we firmly in the era of fully native signal routing?

I love to hear others practical experiences.

 

[Per Boysen]

I always use a 128 sample's buffer, if not mixing under heavy CPU usage. Mostly I record active latency compensation, when monitoring through Logic. Sometimes I monitor by direct output (through the audio interface, not going through Logic) or plain acoustically, which is about the same. I can't say any method has proven superior with me; every situation may call for a different solution in order to maximize the musical playability with the system. The key concept is to have the performer play well and if that means recording with latency I set it up like that and move parts manually after having recorded.

 

[Orren Merton]

I find that 32 samples is too tight when running large songs (large = 60+ tracks at 24/44.1 w/medium amount of plug-ins). But like Per, I find 128 is very solid performance, and low enough that it still doesn't hinder performance.

Orren

 

[Ru_C]

I find that 32 samples is too tight when running large songs (large = 60+ tracks at 24/44.1 w/medium amount of plug-ins). But like Per, I find 128 is very solid performance, and low enough that it still doesn't hinder performance.

Orren

What machine/interface are you on out of interest?

 

[Orren Merton]

Vintage Mac Pro (2006; 4 core 2.66GHz), Mobile I/O ULN8. 32 samples works until the track count gets large, then I start running into overloads.

Orren

 

[Ru_C]

Vintage Mac Pro (2006; 4 core 2.66GHz), Mobile I/O ULN8. 32 samples works until the track count gets large, then I start running into overloads.

Orren

Ok interesting. One of the first tests i did, was to load up one of the demo tracks (the lily Allen one) , & see if the machine could cope with a 32 sample buffer guitar overdub, using a supplied amp sim plugin.

I clicked on 'low latency mode', & it handled it fine with a £100 USB interface (Alesis io2) on a new iMac. I was surprised to say the least !

 

[Orren Merton]

The real trick is, when you have 60+ tracks of audio, 2-4 tracks of software instruments, and 2-3 tracks in record mode, with maybe 30-40 audio effects spread around, all at the same time. If you don't see that happening, then you're golden. But if that's realistic, it's something to keep in mind. Also, some instruments like Omnisphere, Absynth, etc. are known for really taking a lot of CPU cycles.

Orren

 

[Ru_C]

The real trick is, when you have 60+ tracks of audio, 2-4 tracks of software instruments, and 2-3 tracks in record mode, with maybe 30-40 audio effects spread around, all at the same time. If you don't see that happening, then you're golden. But if that's realistic, it's something to keep in mind. Also, some instruments like Omnisphere, Absynth, etc. are known for really taking a lot of CPU cycles.

Orren

Ok right. I look forward to pushing my future Logic Rigs

For the stuff i'm doing though, it's mostly live instruments, tracking 5 piece bands, so i'm interested in setting up multiple pre-fader sends per tracks, for individual headphone mixes...hence the v.low latency requirements.

 

[Orren Merton]

For the stuff i'm doing though, it's mostly live instruments, tracking 5 piece bands, so i'm interested in setting up multiple pre-fader sends per tracks, for individual headphone mixes...hence the v.low latency requirements.

Sounds like the 32 buffer will work brilliantly for you. How are you recording 5-piece bands with an Alesis IO2? Or is that just a plan for the future? My advice is to get a FW interface, as they do better with multi-I/O in general.

Orren

 

[HKC]

But beware of the "sudden death". That's what we call it when people run with too low buffer range and the recording suddenly turns into white noise. I don't hear much talk about it anymore so it may be a thing of the past but on my ancient setup it still happens now and then when I multitrack (16 tracks at once).
Actually I was going to post a thread where I would like to ask if this still happens with more recent setups. We're talking more highend soundcards like RME Raydat, Apogee Symphony etc...... And did anybody ever find out what causes it. I know that lowering the latency cures it but normally too low latency gives clicks and pops not a devestating white noise.

PS I record many 4-5 piece bands as well so I am very interested in this thread being about to upgrade my system.

 

[Ru_C]

For the stuff i'm doing though, it's mostly live instruments, tracking 5 piece bands, so i'm interested in setting up multiple pre-fader sends per tracks, for individual headphone mixes...hence the v.low latency requirements.

Sounds like the 32 buffer will work brilliantly for you. How are you recording 5-piece bands with an Alesis IO2? Or is that just a plan for the future? My advice is to get a FW interface, as they do better with multi-I/O in general.

Orren

For sure, the io2 just happened to be on the rig i tested it with first. I was impressed enough to go out and buy the bigger io26 - really nice little interface with 8 pres for the price.

It has two additional ADAT ins, so I can scale up to 24 ins with some presonus digimax units or suchlike.

I'm planning on installing this in a Rehearsal space, so I can do quick & dirty demos for bands, that can only afford a day.

x

 

[Ru_C]

But beware of the "sudden death". That's what we call it when people run with too low buffer range and the recording suddenly turns into white noise. I don't hear much talk about it anymore so it may be a thing of the past but on my ancient setup it still happens now and then when I multitrack (16 tracks at once).
.

Blimey ! I'll def keep my eye/ears open for this.

 

[Howard W]

But beware of the "sudden death". That's what we call it when people run with too low buffer range and the recording suddenly turns into white noise. I don't hear much talk about it anymore so it may be a thing of the past but on my ancient setup it still happens now and then when I multitrack (16 tracks at once).
Actually I was going to post a thread where I would like to ask if this still happens with more recent setups.

It still happens occasionally, here's a related thread from this forum.

https://logic-users-group.com/forum...distortion-recorded-with-logic-and-duet.2704/

Howard

 

[jonperl]

...

I clicked on 'low latency mode', & it handled it fine with a £100 USB interface (Alesis io2) on a new iMac. I was surprised to say the least !

...

"Low Latency Mode" deals specifically with Plug-In Delay Compensation (has nothin gto do with Buffer settings and I/O latency); namely when it is set to "ALL" in preferences (which is now the Factory Default), you could be screwed when recording overdubs when you have high latency plugs on Tracks, Auxes or Outputs. Have Low Latency as a button on your Transport and keep it on - when you go into Record, it will temporarily disable plugs that cause the delay to exceed your specified minimum...

 

[Ru_C]

...

I clicked on 'low latency mode', & it handled it fine with a £100 USB interface (Alesis io2) on a new iMac. I was surprised to say the least !

...

"Low Latency Mode" deals specifically with Plug-In Delay Compensation (has nothin gto do with Buffer settings and I/O latency); namely when it is set to "ALL" in preferences (which is now the Factory Default), you could be screwed when recording overdubs when you have high latency plugs on Tracks, Auxes or Outputs. Have Low Latency as a button on your Transport and keep it on - when you go into Record, it will temporarily disable plugs that cause the delay to exceed your specified minimum...

Yep, sounds good. I mean PTHD has issues all of it's own when overdubbing with ADC/PDC switched on.

Folks gripe about how it works on all the software, but realistically, all the DAW makers are having to deal with the laws of physics !