4 ms·

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 implem

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.

PaulDavisThe1st 4y ago
> 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.
- chaosprint 4y ago
  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 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.
  - PaulDavisThe1st 4y ago
    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).
jcelerier 4y ago
> 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.
- skybrian 4y ago
  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?
  - PaulDavisThe1st 4y ago
    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.
    - deleted 4y ago
      [deleted]
    - skybrian 4y ago
      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.

PaulDavisThe1st 4y ago

> 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.

chaosprint 4y ago
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 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.
- PaulDavisThe1st 4y ago
  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).

jcelerier 4y ago

> 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.

skybrian 4y ago
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?
- PaulDavisThe1st 4y ago
  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.
  - deleted 4y ago
    [deleted]
  - skybrian 4y ago
    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.