[fractalf]

Why do you want/need a PCI based card? Most audio interfaces run off USB these days and Linux supports almost all USB audio class-compliant cards.. Personally I use NI Komplete Audio 1 and Behringer UMC 1820 (yes, on Linux with pipewire and Bitwig you can use multiple cards!)

[louthy]

> Why do you want/need a PCI based card?

Because I don’t know of any USB device that can do 96 channels of pro IO. Also, it always seemed preferable to me to avoid the issue of ‘yet another wire’ and ‘yet another source of latency’.

Being attached to the PCI bus always felt like the most robust approach to getting audio in with the lowest latency (which it is on Windows).

I’d probably prefer to look at Dante/AES67 before USB, unless there’s a compelling reason not to. Mostly because it gives room for future expansion where a fixed soundcard doesn’t.

> yes, on Linux with pipewire and Bitwig you can use multiple cards

How are you clocking the cards? If they don’t share a single clock then you’re gonna get phase issues. The idea of going back to a word-clock does not appeal to me! I’d much prefer a single device with one clock. The fewer moving parts, the better.

[fractalf]

> Because I don’t know of any USB device that can do 96 channels of pro IO

Holy crap, that's a lot of input channels, didn't know it was possible. I also have a Behringer ADA8200 that connect to the 1820 (ADAT) which gives me 16 inputs. I'll admit it's not enough to run all of my gear in at once and this is just a home studio. I can imagine 96 channels is very useful. Maybe 6x my setup would work = 96 inputs ;)

> How are you clocking the cards?

You are obviously more advanced than me, not sure what you mean here. I'm using the NI card basically just for the output and the Behringer for inputs, 24/48. When recording, if there is a latency (which normally Bitwig can "fix") it's just a matter of moving the audio a couple of ms.

> PCI

I've had earlier problems with PCI cards, in that they pick up a lot of internal noise from the motherboard, so I've stayed away from that after firewire/usb cards became available

[louthy]

> Holy crap, that's a lot of input channels

96 output too. I have a lot of outboard gear, my SSL Sigma Delta alone is 32 in/out for mixing, that's before I get onto all my outboard compressors, EQs, reverbs, synths, drum machines, and modular gear. I use the soundcard pretty much as a virtual patch-bay, I can send and return to any piece of gear in my studio like a plugin (get Bitwig to measure the latency and automagically everything syncs).

> You are obviously more advanced than me, not sure what you mean here.

Because digital audio is in discrete time-slices, you need to sync the slice starts so all devices are aligned. Otherwise you can have subframe issues where phasing becomes a problem - in the worst cases it can cause audio glitches (which I've had with older AD/DA convertors).

This can be solved by using a Word Clock [1], which is the 'old' way, or via AES/MADI/Dante - which all have their own clocking protocols. The devices all need to be clocked by the same source, which you can't do by pairing devices in software.

My current soundcard is connected, via MADI, to my 3xFerrofish-A32-Pro AD/DA units. The soundcard is the source of truth for timing. The AD/DA devices all sync to the soundcard and therefore have no phasing issues. This gives 96 channels of analogue audio in and 96 channels of analogue audio out.

> I've had earlier problems with PCI cards, in that they pick up a lot of internal noise from the motherboard

Sure, if you plug analogue sources directly into the device. But my analogue inputs/outputs are all going into my AD/DA convertors, which are separate units. They are digitising the audio and by the time it arrives (via MADI) into my soundcard it's already digital. That means no chance of any noise from the computer.

Probably what I need is something like this [2], it seems to have Linux support, but unfortunately isn't produced any more! I seem to find that a lot with these Dante devices. Maybe I'd be better with AES67 as that's an open-standard that is compatible with Dante, just I haven't found anything yet!

In theory (because Dante/AES67 is over IP) a virtual soundcard can be written, I certainly have a powerful enough machine to run it, but again I haven't found anything yet. I'd write it myself, but I have a million things going on, and that's a distraction I don't need!

It does seem like there's a bit of a gap when you get to the real pro end of audio, which is a shame because the stability of Linux would is a huuuuuge selling point when producing music.

[1] https://en.wikipedia.org/wiki/Word_clock

[2] https://www.audinate.com/products/dante-enabled/four-audio/f...

[fractalf]

Hey man, thanks for elaborating, I do understand the limits you are presenting much better now. Aaand, this actually makes me want one of those cards myself! I could just have one of those PCI cards and 3-4 of the Behringer ADA8200 and I could run ALL of my synths into my PC.

[louthy]

> Aaand, this actually makes me want one of those cards myself!

heh, sorry!

Looks like I probably only searched for "Dante virtual-soundcard linux" in the past, because when I search for "AES67 virtual soundcard linux" I got [1] and [2] which may be the answer to my problems (if my machine can handle that many audio streams, but it should be able to with 64 cores haha!).

Looks like that's going to be my project tomorrow.

[1] https://github.com/bondagit/aes67-linux-daemon

[2] https://gitlab.freedesktop.org/pipewire/pipewire/-/wikis/AES...

[fractalf]

Oh, nice, very interesting! I found you on mastadon, please write about your experience, I'm curious now :)