Maybe the web was initially not designed for doing these kinds of things, but on the other hand it is basically the only platform that enables people to click a link and immediately start using whatever app is behind the URL. That's fundamentally why the web has grown as much as it has, and why there is huge incentive for people to build more APIs that let people do more with that platform.

I suppose you could in theory build a similar kind of platform that allows you to the same things in better ways. There may even be a lot of money there. But it hasn't happened, and so it makes a lot of sense IMO to build these kinds of apps.

On the third hand: distribution.

An app that runs in the browser offers incredible advantages over an app that you have to download and install on your specific machine/OS.

A DAW in the browser sounds insane, but the same was said about everything: word processors, spreadsheets, maps, and look where we are now.

I never worried about milliseconds of delay in spreadsheets, but in a DAW I do. Things are not all the same.

I think it goes way beyond toys. Here's my use case: I'm building an audio sample pipeline. And I want people to be able to preview combinations of loops and samples in the browser. Asking someone to download a native app just to preview is not awesome: https://signalsandsorcery.org

IMHO, your usage of Tone.js may not be a strong enough argument for your point. Tone.js is simply too high-level to offer the features mentioned here.

Not sure what you mean? It's already serving my use case isn't it?

Good point.

I think you do accept the convenience of browsers (many cool projects such as this one https://learningsynths.ableton.com/).

So now the problem is the audio performance.

As I post in the comments below, we now have WASM, so C++ and Rust can all run in browsers. This can provide a near-native audio performance.

Just take Glicol, the live coding language I design as an example: it runs in browsers (https://glicol.org) and it also runs as a VST plugin (https://youtu.be/tmmBhBmIEW0), or you can use the audio engine to write VST plugin (https://github.com/chaosprint/dattorro-vst-rs).

> As I post in the comments below, we now have WASM, so C++ and Rust can all run in browsers. This can provide a near-native audio performance.

Native audio apps require realtime scheduling and memory locking, things you cannot do/control in the browser.

The problem is that you can likely get 70-85% of "it" done in the browser, but when the user/performer need the remaining 15-30% for whatever reason, what do they do then?

Very rarely do you get that level of control even on modern systems. People aren't making music on ASIO-capable sound cards, they're using ASIO4All which implements ASIO on top of your boring MME/WASAPI audio stack. On any modern OS, you don't write directly to the sound card, you write to a buffer which the system mixer does the FX graph on, then that goes to the underlying sound system. CoreAudio, WASAPI and PipeWire are all built like this.

Realtime threads are attempted to be solved by AudioWorklet which allow a high-priority thread.

We probably won't get to the level of 100% assembly programming, but we can get close, and I think that's fine.

Where I get upset at Web Audio is the amount of time spent in all the oscillators and FX nodes; I think you're better off ignoring those just because of how limiting and awkward they are.

> People aren't making music on ASIO-capable sound cards,

sorry what :) I don't know anyone who does a bit of remotely serious music making who doesn't have at least some USB focusrite or something like that

> On any modern OS, you don't write directly to the sound card, you write to a buffer which the system mixer does the FX graph on, then that goes to the underlying sound system. CoreAudio, WASAPI and PipeWire are all built like this.

Paul Davis is the author of JACK and Ardour fyi, I think they have a good idea of how things work :p buffers are unavoidable but I don't see webaudio allowing me to use a 64 frames buffer size and still be able to put in some effects and play with some softsynths like I have right now, or even being able to run isolcpu and tweak DMAs to entirely devote specific CPU cores to audio processing.

For "people" read amateurs who haven't even heard of Focusrite (I hadn't.)

I expect many people fooling around with GarageBand or VCV Rack don't have any special sound card. Also, it doesn't seem that uncommon to for musicians to make recordings using a cell phone?

Nobody makes finished recordings for distribution with a cell phone. Field recordings for use as samples? Sure. Something to get the overall feel of a performance? Sure. A dedicated mic on an instrument because nothing else was available? Sure.

But recording when it matters is dependent on the microphone, the pre-amp and the A/D converter, none of which are of suitable quality in a cell phone to be the basis for a "serious" recording.

This is not what I see on YouTube. Professional recordings are outnumbered by amateur ones. They are as "finished" as they're going to be and sometimes they get a big audience.

Sure, maybe most professionals don't do that but you said nobody does that, as if amateurs don't exist.

Well, I guess there's also recordings with a line-in from a mixing console with the mics&pre-amps on the other side of that, and sure, that's a thing, and I ignored that with my comments. That setup gives you just-about-good-enough quality (certainly way better than ye olde "direct from desk" bootlegs of past time).

I was thinking more of setups where you connect a mic/pre directly into the phone.

Most of the time I've seen people doing this, however, they are using "native" recording apps on their phones, not a browser.

> Very rarely do you get that level of control even on modern systems.

Windows, macOS, Linux all offer this control.

AudioWorklet does not offer much control of thread priority (read the source code of any DAW to see how much is actually done), and also isn't clearly appropriate for parallelization of DAW processing, which is a standard architecture in any current DAW.

It's not really about asm programming. It's about the ability with the host OS in all the appropriate and required ways. Browsers are extremely unlikely to permit this, for a variety of very sensible reasons.

I think it's all about trade-offs.

For example, I can also point out that the latency on Windows, macOS are too high as these are general-purpose oriented systems. See this paper:

http://eecs.qmul.ac.uk/~andrewm/mcpherson_nime2016.pdf

Similar dilemma we have for the general-purposed browsers. To push to a limit, we can only resort to bare metal or Xenomai Linux like Daisy or Bela. But for the same reason we use Windows and macOS, we trade off some audio performance for a better overall experience.

Most of the problems with the lower limit on audio latency these days come down to hardware (motherboard/PCI bus level) rather than the OS. Both macOS and Linux can routinely handle values significantly lower than the 10msec mentioned in the paper you cited, given the right hardware (more or less guaranteed for macOS, much harder to ensure for Linux).

Yep. Pure Data and SuperCollider can also run in the browser.

But there's a difference between C/C++/Rust audio apps compiled to WASM on the one hand and writing audio code in JS on the other hand.

Then again, I think it's perfectly ok for people to experiment with audio in browser. "Native" audio desktop applications will always be superior, so there's nothing to be afraid of.