I mean, I have currently 5 soundcards on my desktop computer, 1 pro over PCIe, 2 pro over USB, 1 standard HD Audio and 1 output in my screen's HDMI, and I can assure you that even on windows it's definitely not a smooth ride between WASAPI, ASIO, MME, WDMKS... things crash or get stuck routinely, and I sometimes get random loud buzzes or no sound until I reboot. With my band we had tons of issues with a M-Audio card on windows fixating itself on the wrong sampling rate as soon as a desktop music playing was launched (this includes web browsers), entirely preventing playback on Ableton Live for instance.
It's barely better on macOS, e.g. look at that shit: https://github.com/OSSIA/score/issues/778. macOS also sacrifices some low-latency when comparing the same hardware on it versus Linux (with raw ALSA or JACK) and Windows with ASIO.
well, no, the web audio API & stuff is explicitly being sold as something that will be equal in capability to other solutions, so definitely things used for pro audio.
It seems like a relatively contained problem to take multiple streams of PCM audio into a computer, locked to word clock. Then the hard, fiddly aspects of analog design can be left to high-quality outboard converters which speak MADI or whatever.
I say "contained" because I don't want to minimize the difficulty of getting such a design right, but it seems like you only have to get it right once.
Then we don't need a USB driver for each converter! We only need one USB driver, for the digital i/o interface. And we can polish the driver for that one interface until it's actually reliable, instead of relying on the sketchy one-off driver for this year's soon-to-be-obsolete USB audio interface.
ETA: Many such converters (most of them high-end) listed here: https://www.sweetwater.com/c796--AD_DA_Converters
If I'm not mistaken the underlying transport protocol is the same for all of those - you can get a coaxial to XLR to get S/PDIF into an AES port or a coaxial to TOSLink converter to get S/PDIF data into an ADAT port.
You can look into the Madiface XT maybe ? https://www.rme-audio.de/hdspe-madi-fx.html or the other RME products which have enough digital I/O to cover a lot of needs
But the point is that I want open source drivers, and open source hardware! I don't want to depend on the health and the priorities of a commercial entity. I want to be able to inspect the driver software and contribute towards perfecting it!
Compared to, say, GPUs, the needs of multichannel digital audio i/o are modest and not changing very much over time.
Some quick usage of Wikipedia suggest that you are, sadly, mistaken. ADAT lightpipe seems to use a completely different protocol (more capable?), and while S/PDIF and AES/EBU use quite similar protocols, they differ in impedance and max and min voltages, so I'm guessing that directly electrically connecting the two controllers is a bad idea for both the controller and the data.
(Licensing could be a concern though if ADAT lightpipe is proprietary — I'm not clear on that. But then we could just use other open protocols instead.)
Windows probably has the best audio stack of them all (battle tested, easy to use UX). In comparison, JACK/pulse are pretty good as well and people use it professionally. Like most things on Linux, the initial UX is a bit of a pain, but once you get past it you get amazing customization, even more so than Windows.
These are the only abstractions I'm aware of on Linux systems - and they are simply built on top of ALSA which is the kernel's core sound API. Almost all desktop environments simply install PulseAudio, and you never have to worry about sound.
CoreAudio is far and away the most impressive and powerful audio API across the board. It's ridiculous how much you can do with it and the abstractions it provides.
It's also been better, faster, and supported lower latency than Windows up until WASAPI came around - you needed to use ASIO on Windows for years. Even today you can't programmatically change the sample rate of a device on Windows, which means that engines like RtAudio have to do resampling under the hood to support basic functionality that's been in CoreAudio.
The only problem with it is that the documentation has evaporated. Which admittedly is a big one.
New MBP 16". Apparently it's a problem with the T2 security chip and affects almost all external USB 2.0/3.0 audio devices. Just do a google search for audio drop outs and T2. All digital audio goes through the T2 chip which seems to add latency and these drop outs. I think the only solution at the moment is to use a USB-C native device (but I haven't tested this). If you already have substantial hardware like I do, Catalina is basically unusable (and no fix in sight from Apple...).
I understand with brand new machines it's a problem, but buying less-than battle tested hardware/software has always been problematic. If you already have substantial hardware, you probably shouldn't be buying a brand new machine and expecting it to just work after a major OS upgrade.
And I've been burned enough times by OS upgrades that I know to be cautious. I'm not saying the situation is good, just that it's a shitshow everywhere. I got around a dozen emails from my device and software vendors telling me not to use Catalina, so I'm not yearning to go out and buy a new Mac to run my software on and connect my hardware to it. I'll shit on Macs for that, but under the hood, CoreAudio is much more impressive than anything on Windows or Linux.
(Nitpick: Linux has nothing to do with Pulseaudio, it's just the kernel.)
Easily, reliably, and quickly playing audio is table stakes for Windows and Mac OS. They've been doing it for years. But instead of improving one or two audio APIs, on Linux it is time for a new audio server that will, but doesn't, solve all of the problems of every predecessor.