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Can two USB audio interfaces be aggregated together, with the intention of using all its outputs to connect to the inputs of 12 or 20 track analog mixers that are usually used for live-sound? Secondly, would the USB connection be sufficient to synchronize clocks, or will SPDIF connections be needed to improve synchronization?

This is in contrast to a few available usb interfaces that already do this via ADDA's on each channel strip, for analog mixing in the desk. For example, the soundcraft MTK.

I currently have a 2014 Focusrite 6i6. I'm looking aggregate it with another multi-output usb interface like the Behringer FCA610 (eight 1/4" outputs) and try to patch connections with a Yamaha MG16XU and use its circuitry to mix songs.

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  • What OS? I'd imagine this is pretty easy on Mac, though you might have to buy software. idk for Win.
    – Tetsujin
    May 11 at 17:35
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    @tetsujin you do not even consider GNU/Linux :"(?
    – Tom
    May 11 at 17:55
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    @Tom - nope. Know absolutely nothing about it;) i know little enough about Windows for audio routing, all I do know is it's.. not exactly user-friendly.
    – Tetsujin
    May 11 at 17:58
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    Joke apart, if you're on Linux it might be possible: jackaudio.org/faq/multiple_devices.html
    – Tom
    May 11 at 19:25
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    @leftaroundabout it's being done for multiple inputs. There's zero content online that uses the technique as multiple outputs. I'm using an old Reason DAW, so I don't have plugins, nor was ever attracted by them. I'm lookin to explore the hybrid analog technique. I've tried it on a smaller mixer, and it has had great results in mixing 3 stereo busses.
    – wearashirt
    May 12 at 3:26
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Here is an old Sound on Sound article: https://www.soundonsound.com/techniques/using-multiple-audio-interfaces-together

The article is 15 years old, but the basic problem has not changed at all: every D/A (and A/D) converter that you'll need for turning a digital signal to an analog signal (or vice versa) uses a clock called word clock that makes the converter "tick" at a steady rate. The steadiness of this clock pulse (i.e. low audio clock jitter) is very important for sound quality in digital audio systems.

If you have multiple audio interfaces, you'll have multiple clocks timing their D/A (and A/D) converters. Each clock can either be synchronized to an external master work clock, or it can run independently at its own speed. If you have multiple unsynchronized clocks, even if they all produce a nominal 44100 Hz or 48000 Hz rate, they will run at slightly different speeds, meaning that at some point in time one of the clocks has ticked one more time than the other clock. And then another tick. And another. And at some point, let's say, audio interface A has output 1000 more samples than audio interface B. But all of the samples are coming from the same source DAW program (Cubase, Ableton, etc.) - what to do?

Let's consider all the software running on the DAW computer as a single "program", even though it consists of multiple parts, including the operating system, device drivers, and application software. Conceptually, the software can resort to one or more of the following methods to manage the problematic situation:

  • Tell interface A when it asks for more samples: "I don't have any more samples for you. You'll have to make up stuff, or output silence, or repeat whatever the last buffer was, or something. Sorry!" Interface A's clock will keep ticking and its D/A converter will keep producing an analog signal, regardless of whether it was given data by the computer or not. --> You'll get artifacts in the sound.
  • Tell interface B: "Why are you so slow? I have a whole new buffer of samples ready for you, but you haven't even consumed the previous one yet!? This other interface has been able to consume everything I've given to it, but not you. I'm going to throw the data away!" --> You'll get artifacts in the sound.
  • Create a virtual "audio clock speed clutch" which gradually slows down or speeds up audio, essentially pitch-shifting and/or time-stretching audio so that on average, all the different audio interfaces stay roughly within the same time window. --> You'll get artifacts in the sound.

These problems can be avoided by having a master word clock in the system. For example so that audio interface A works as the master and has a word clock output, and interface B is a slave with a word clock input. But not every audio interface has these facilities.

If the word clocks are not synchronized, there will be artifacts. What kind of artifacts, how much and are they a problem - you'll have to try and see. Thanks to @ojs, I found that USB audio devices are supposed to somehow synchronize their word clocks to the 1 ms (1000 Hz) bus clock rate. Maybe this works nicely and without artifacts, with the devices that you have.

After the Sound on Sound article was written, a new type of audio device has emerged, the USB microphone, which means that each microphone has a built-in audio interface, A/D conversion and a _word clock. Not getting total word clock sync chaos relies on USB bus clock sync.

I'm not sure how Apple's aggregate audio device and various other similar systems work, but there has to be some kind of a compromise with audio quality/artifacts and latency. There's a "drift correction" feature in the Apple thingy, which tries to keep the average speed in control. Probably just fine for casual music consumption and many other uses. But for some uses such a system might not be OK. Using an aggregate device for mixing, if there are no word clock input/output facilities, might be on the NOT OK side. You'll have to test if it's good enough for your purposes.

Whether this works well enough for your needs or not, depends on your needs, and on the exact combination of hardware and software and their settings. It might be doable with some combination of products and settings, for some uses.

Anyway, an application program such as Cubase or Ableton, wants to see just one output audio interface, and that's why a virtual "aggregate device" is needed. On the Mac, the operating system's aggregate device is reported to work well for multiple USB devices (see here for an example), but on Windows, something like ASIO4ALL is needed to create such an aggregate device https://www.sweetwater.com/sweetcare/articles/aggregate-audio-devices-and-drivers-for-pc-and-mac/#ASIO4ALL-for-Windows

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    For USB devices running in in synchronous mode (like audio interfaces marketed for musicians tend to do) there's the USB bus clock that could be used to synchronize the word clocks of different USB devices.
    – ojs
    May 15 at 14:05
  • @ojs Thanks for the keyword, I found stuff explaining the different modes, for example usb.org/sites/default/files/audio10.pdf (from 1998) In synchronous mode, each device adjusts its clock to the bus clock's 1 ms tick. Anyway, from the point of view of the aggregate driver that sits between the audio interfaces and an application program, audio interfaces running at different and drifting speeds has to be taken into account anyway. Some devices might not be on the same bus, not USB at all, or maybe they just don't want to cooperate. Maybe it works, maybe it doesn't. May 15 at 14:34
  • To me the biggest surprise was that at least entry-level musician-oriented USB cards use synchronous mode at all, because in synchronous mode they are at the mercy of USB clock that is not necessarily very accurate and asynchronous or adaptive modes would allow running their own clocks. You're right that in general case synchronizing between different interfaces needs external clock or time stretching. For synchronous mode USB interfaces the bus clock can work as the external clock.
    – ojs
    May 15 at 15:37
  • @ojs The more I read about this, the more I'm thinking, this might not always be completely problem-free... ;) I'm sticking with a single audio interface that has enough inputs and outputs. May 15 at 16:22
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In some cases you can use two USB audio devices at the same time through an OS software only. MacOS does allow interface aggregation. The ASIO4All Windows driver supports aggregation on Windows. I'm not sure about the Linux drivers, but there is probably some low latency audio drivers available that would do it.

Usually multiple devices are synced on the hardware side before sending into and out of the DAW, and if you are trying to use multiple units for simultaneous output they would have to be synced in some way. Using the software to aggregate may give you consistency problems such as drop outs, noise, or clock errors.

If your hardware supports S/PDIF or ADAT sync, then you can sync the units so that their outputs will be simultaneous.

There are people that prefer to mix through their analogue boards, especially vintage hardware such as the type that contains the Neve channel strips or other boutique hardware. Software plugins are pretty good at emulating tube pre-amps and compression circuitry etc., but there is arguably a different response sending a signal through a physical tube or tape compressor.

Many of the digital interfaces are designed to send and receive the channels for engineers that want to use their favorite hardware in their processes.

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