Debian has stopped supporting x86 32bits recently, Chrome did so 9 years or so ago.
We've carefully ran some numbers before doing this, and this affects a few hundreds to a few thousand people (hard to say, ballpark), and most of those people are on 64bits CPUs, but are using a 32bits Firefox or 32bits userspace.
The comparatively high ratio of 32bits users on Windows is not naively applicable to the Linux Desktop population, that has migrated ages ago.
Those methods are sub-sample accurate, granted you call them a bit in advance to account for the cross-thread communication, as you say. But yes, in general this was designed (prior to me becoming an editor) with scheduling in mind, not with low-latency interactivity. That said, it goes quite far.
Other systems go further, such as Web Audio Modules (that builds on top of AudioWorklet) implement sample-accurate parameter change from within the rendering thread, using wait-free ring-buffers. That requires `SharedArrayBuffer` but works great, and is the lowest latency possible (since it uses atomic loads and stores from e.g. the main thread to the rendering thread).
`jj commit -i` (or a lot of commands `-i`) and maybe `snapshot.auto-track="none()"` in the config, to a certain extent is what I use. I used to do the same with mercurial. In practice, I also use `absorb` a lot, that leaves unrelated files and chunks alone.
He did last year, alongside mayor Hidalgo and others, such as Tony Estanguet, who is a former athlete and was overseeing the olympics and the minister for sports.
You're out of luck on Safari because it seems that the important APIs aren't implemented yet: https://developer.mozilla.org/en-US/docs/Web/API/AudioContex... (scroll down). This means this cannot be made correct, e.g. there will always be a desynchronization when using high latency audio output devices such as anything wireless. Their handhelds (on which non Safari is not allowed) and their MacBooks, in wired/built-in speaker mode have excellent latency figures and we can get away with not doing anything explicit there, granted the audio is not happening somehow before the visuals, that is jarring for a human. A bit late is a lot more natural if it can't be exact.
lmk if you need further details on tight synchronization of real time audio and visuals, [email protected], happy to help, and congratulations on the delightful websites!
This is planned, and important, and we'll fix it hopefully soon, it's long overdue. I'm sorry this hasn't happened yet, it's always a game of priorities that can never satisfy everybody on time. It however ranks fairly high on my personal list.
As one could imagine it's a bit (read: a lot) more complicated than just pausing the AudioContext after some time of silence, but we'll get it fixed regardless, it's possible because others did it. There are tradeoffs we're willing to do.
Source: Firefox implementer of a lot of things around this, editor of the Web Audio API standard.
It does not, it serves no purpose for only listening to music.
Producing, broadcasting, or any sort of scenario that require reliable low latency can benefit from a real time kernel, but not merely listening to music.
Indeed. I work on the Firefox media stack and we have been grabbing wake locks when video playback is happening for a long time. Occasionally e.g. on some linux desktop variant this has malfunctioned and we're alerted in no time and fix it.
The Wake Lock API is for other use cases, such as recipe websites, or other document for which you don't want the screen to go away / dim, the kind where you happen to need to look at the screen for long period of time without touching it/interacting w/ mouse and keyboard.
Prior to this API being introduced, websites used to play an inaudible/almost invisible looping media file to keep the screen awake. This has power usage implication, a small single digit number of watts (1 to 3.5 depending on os, hardware, mobile or not) is required to keep audio running (because of high priority threads, frequent wakeups, and simply because the audio hardware itself needs power).
Yes, I use rr all day every day (to record Firefox executions) on a rather recent Threaripper Pro 7950, and also with Pernosco. The rr wiki on GitHub explains how to make it work. Once the small workaround is in place it works very reliably.
The way this works (and I'm obviously taking a high level view here) is by comparing what is being played to what is being captured. There is an inherent latency in between what is called the capture stream (the mic) and the reverse stream (what is being output to the speakers, be it people taking or music or whatever), and by finding this latency and comparing, one can cancel the music from the speech captured.
Within a single process, or tree of processes that can cooperate, this is straightforward (modulo the actual audio signal processing which isn't) to do: keep what you're playing for a few hundreds milliseconds around, compare to what you're getting in the microphone, find correlations, cancel.
If the process aren't related there are multiple ways to do this. Either the OS provides a capture API that does the cancellation, this is what happens e.g. on macOS for Firefox and Safari, you can use this. The OS knows what is being output. This is often available on mobile as well.
Sometimes (Linux desktop, Windows) the OS provides a loopback stream: a way to capture the audio that is being played back, and that can similarly be used for cancellation.
If none of this is available, you mix the audio output and perform cancellation yourself, and the behaviour your observe happens.
Source: I do that, but at Mozilla and we unsurprisingly have the same problems and solutions.
We've been working on the Web Codecs API for a few years now. It only handles encoding and decoding media codecs, ffmpeg does much more (container handling, filters and everything needed really).
Web Codecs can take a compressed media packet and get you the decoded image or audio buffer it corresponds to. Conversely it can take audio or images with timestamps and get you a series of encoded media packets you can then containerize (we say mux) and get you e.g. an mp4 file.
I (Firefox developer working on anything media related) got in contact with the dev on Twitter, and he told me that Web Codecs was missing (and we're shipping this in a month or so, it's been in Nightly for some time), and something to save project file to disk (https://developer.mozilla.org/en-US/docs/Web/API/Window/show...).
So I spoofed the user-agent in a nightly build here on my Linux desktop workstation, then had to alias one method that we should have implemented years ago but only have with a `moz` prefix (`HTMLMediaElement.mozCaptureStream`). This is on us to fix.
Then it looks like a worker script is served with the `Content-Type` `text/html` instead of `application/javascript` or something like that. We also have a pref flip to bypass that check, so I did that, but this is on the dev to fix.
When you do this it works, I've loaded project demos containing videos, audio, various things composited on top, scrubbed the timeline aggressively in a debug build, moved things around in various bits of the interface and also in the rendering frame, etc., things seem to work as they should, perf is as I'd expect it to be (and again, I'm running it in a debug build with optimizations disabled for anything media related, enabled for other parts of the browser).
What's missing is `window.showSaveFilePicker` and file system related stuff. It's possible to use https://developer.mozilla.org/en-US/docs/Web/API/File_System... instead (that we ship, e.g. Photoshop on the Web uses it). We think that it's much less scary than giving access to the file system to a content process of a Web browser. Maybe because videos can sometimes be extremely big files, direct access to the FS could be of use there. Thankfully, we also ship extremely modern video encoders to make them tiny instead, but that's currently a limitation Firefox has, for better or worse.
We're aiming to release this half, so in a month or so, on all desktop platforms, mobile will follow shortly after.
Then we'll gradually optimize (e.g. enable the use of more hardware encoders, decoding being generally in hardware at launch of supported) release after release, but generally almost everything will be supported at launch.
about:preferences#privacy, scroll down a bit, it's under "Permissions". You can also adjust it when on the site, using the icon at the left of the URL bar.