Has anyone had success recording into Studio One 4 with a Quantum and not used Low Z Monitoring? For being a fast interface I find there is stil too much latency to record without using Low Z.
Low-z only means using the lowest buffersize you have adjusted - without Low-z it uses the buffersize set for dropout-protection which can be very high.
So it doesn't make any sense not using Low-z when recording.
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Softube Console1 MK II, Volume 2
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Thanks for the reply,
The reason I ask, is that I am using Softube Console One and you cannot even have it on a track whether is enabled or not enabled and use low Z on the master bus when you enable record. I only use console one for mixing but its nice to have it all set up on a template.
The issue is that Console One plugins are latent (add additional latency).
That latency has to be compensated (whether the plugin is turned on/off), so you can immediately mute/unmute the track/bus.
If latency compensation wasn't enabled for plugins that are turned off, it would have to be calculated/applied after you hit the transport (causing sluggish response).
Too much latency? Is that only when you use Console One? I'll run some tests but with the Quantum even the regular latency times (no green Z) are listed as much lower than the same buffers were on my last USB interfaces...
Studio One uses a "Hybrid Buffering" technique.
If you have Dropout Protection set Maximum (2048 samples), the extra buffering significantly raises round-trip latency.
Using the Green Z button allows monitoring that particular track/s with normal latency (bypasses the dropout protection buffer for that track/s).
The larger Dropout Protection buffer allows more efficient CPU use (at the cost of additional latency).
The Green Z button allows you to monitor tracks without that additional buffer/latency.
This gives you the best of both worlds (more efficient CPU use... and the ability to work at low latency).
Studio One paired with Quantum is a great low-latency combination.
Studio One is one of the few DAW applications where you can effectively work at 96k using a 32-sample ASIO buffer size. That's 1ms total round-trip latency.
You can effectively play/monitor in realtime thru AmpSim plugins like Helix Native or the new TH-U with super tight timing (1ms total round-trip latency).
CPU use is pretty high with Helix Native... but the audio is glitch-free.
Without Hybrid Buffering, this would be much less practical...
pauldormer1 wroteYou seem rather knowledgeable Mr Roseberry
There's not a "one side fits all" answer.
Each scenario is different.
The Quantum/2 can go down to a 16-sample ASIO buffer size.
No current machine is going to be able to effectively work at that setting.
To effectively work at the 32-sample ASIO buffer size (especially with higher sample-rates), it takes a fast recent-make machine.
Given your machine, I'd say 64-128 sample ASIO buffer size would be a reasonable starting point.
Regarding sample-rate, I do think the sound-stage is a little more 3D at higher sample-rates.
That said, I think there are numerous other factors that play a larger role in the ultimate quality of a recording. Mics/position/instrument/performance/arrangement/song/room/etc all play a larger factor than the sample-rate.
Some folks get hung-up on sample-rate. I had one local guy tell me, "No professional would work at anything other than 96k." Ironically, this person hasn't released anything professionally.
My wife is on the local classic-rock station's morning show. They do everything at 44.1k.
No record has been (or not been) purchased solely based on the sample-rate used to record it.
There are situations where 44.1k is more practical (large/dense projects).
If working with video, 48k is more practical.
If you're recording an acoustic guitar player who simultaneously sings... and you want to capture as much detail as possible, working at higher sample-rates isn't a bad idea (the overall project load isn't very high).
Quantum set to 16 buffers 48k
That is about 1.5 feet between the mic and your speaker for recording general tracks and 5 feet if you need to punch into an existing track.
Yes I think audio-wise the bitrate is more important (although I'm interested in recording some hi res sounds of nature to slow down! Hear what are above our ears). I have tended to stick with 44.1Khz, just for compatibility and familiarities sake really. I was under the impression though that increasing sample rate leads to slightly less latency?
I think you are right 64 samples, I've had a few clicks at 32 when running a lot of processing... I'll try working around these settings, and jpetit that seems a fairly decent there. Why may I ask tho, is LLM not OK for punch ins?
Please read the low latency handbook found in the FAQ section of the forum.
One side affect of LLM is that you cannot hear the existing track prior to a punch in when in that mode.
The handbook explains 3 to 4 work arounds the simplest one of which is just to not use LLM in that situation.
Yes increasing sample rate decreases latency.
It’s all really a matter of what works for you.
For most people anything under 10 ms RTL is sufficient.
jpettit1 is da man!
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