Tie Lines->USB->DAW and back, where to set level going to DAW

[Jline]

Hi, I just set up my SQ-5 as a front end to my DAW (Studio One Pro). I used Tie Lines to send local inputs to USB-B, then routed 32 channels to the DAW in a 1:1 relationship. Out of the DAW back to the SQ via the USB-B as the input source to channels 1-32 of SQ-5. Everything seemed OK except for one problem, the Preamp no longer controls the gain of the mic socket, it only controls the trim function. How do I adjust the mic socket gain so the level is appropriate for my DAW?

[volounteer]

@Jline

I raise the levels in my DAW.
YMMV

[KeithJ A&H]

Hi @Jline,

If you’re wanting to use the DAW ‘in-line’, the easiest way to control socket preamps is by touching the input socket numbers in the I/O screen.
(https://www.allen-heath.com/media/SQ_ReferenceGuide_V1_4_0.pdf#page=20&zoom=160,-237,302).

If you’re not using the DAW for processing however, but do need to monitor channels individually, you might also consider making use of Tie Lines and the Input Channel Patch libraries.
This way, when tracking, you have control over all monitoring, lower latency, and faster preamp/setup control, but when you do want to mix back through the SQ, you can simply switch to the USB input.

If you need more info on this kind of setup, just pop us a message using support.allen-heath.com 🙂

Hope this helps!
Keith.

[Jline]

Hi Keith,

Thanks for the reply. I didn’t realize that socket preamps could be controlled from the I/O screen, that solves the problem!

Regarding recording, I’m not planning on using the DAW for processing during recording. I think I get what you’re saying by using Tie Lines and the Input Channel Patch libraries. To confirm, when recording use Tie Lines to send signal to DAW, but monitor the live signal using the local (or SLink) inputs thereby allowing me to control the preamp from the rotary encoder as well as have minimal latency.
Then, when it is time to listen, simply switch all inputs to USB and monitor back through the board. Correct?

While I could see this working for live recordings and virtual sound checks, this would not work in situations where overdubs are required and you must monitor through the DAW. So far, I’m not sure if I can get the latency down to an acceptable level when monitoring through the DAW. Just curious, what type of latency values have other users typically reported? I realize this is a loaded question as many factors contribute to latency.

[KeithJ A&H]

Hi @jline,

That is correct 🙂
So…

For initial tracking of each source you have:
Input socket -> Input channel -> Mixes/Monitoring (all processing available to use as needed)
and
Same input socket -> Tie Lines -> USB -> Recording (no processing at all)

Then for playback, switch inputs as mentioned:
DAW output to correct USB channel -> USB input -> Input channel -> Mixes/Monitoring (will then use exactly the same processing and mixes)

For overdubbing, you can use a combination of both by only switching the input of the channel you are recording back to the input socket.
This basic setup would ‘limit’ you to working with 32 channels in the DAW, but for more than this you’d only need to do some summing in the DAW.

Regarding latency, it does depend on the hardware, software and buffer settings used.
@64 samples for example, I have a round trip latency of around 4ms, but @4096 samples, this is up near 100ms.
Of course you’d probably experiment to find the lowest stable buffer setting when overdubbing, but are you needing to monitor through the DAW?
Overdubbing with the setup above, you’d only need to find the offset in recording (for your buffer/latency), then shift any overdubs back by that (fixed) amount.
Monitoring of the live signal would still be through the desk, and therefore below 0.7ms and not an issue 🙂

Thanks,
Keith.

[Jline]

Hi Keith,

Thanks again for the response. In regards to your suggestion:

For overdubbing, you can use a combination of both by only switching the input of the channel you are recording back to the input socket.

One problem I perceive with your suggestion is that if I am doing a lot of punching during overdubs (such as working with a singer that is struggling and only singing a section at a time) you really must monitor through the DAW the whole time, otherwise, the artist will not hear their previous track during overdubbing. It seems impractical to constantly switch input sources on the console for the overdub track. I could see this working in a situation where only a few punches are needed, but probably not a session requiring a lot of overdubs.

The smallest buffer size I can utilize before things get unstable is 128 samples at a latency of 9.26 ms. (2018 MacBook Pro, 16 gb RAM, adaptor from USB2 to USB-C ). This is tested with only 16 tracks so far.

I can definitely hear the difference when switching inputs on the SQ between the local socket and the USB return from the DAW. I could probably make 9.26 ms work but it is not ideal especially compared to the latest recording interfaces such as those from Apogee or Universal Audio. Of course, it’s probably not fair to compare latency between an interface on USB2 vs a thunderbolt interface.

Cheers

[Author]

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