It doesn't at the moment, but I think it probably should. There's a non-trivial amount of clock drift that can happen over long periods of time.

Thanks for the heads up! Just added a license to the repo.

Yes! The very next step.

It works on other platforms! Just not as smooth as Chrome.

Yeah the threshold is pretty brutal, but it is enough. Experimentally, I'd say you need under 2-3ms but even at 1ms you can start to hear some phase differences.

Most of the time, I think my synchronization algorithm is actually sub-1ms, but it can be worse depending on unstable network conditions.

How are you measuring this? I'm surprised the Web Audio API scheduling system has that much insight into the hardware latency.

Great question! There's two steps:

First, I do clock synchronization with a central server so that all clients can agree on a time reference.

Then, instead of directly manipulating the hardware audio ring buffers (which browsers don't allow), I use the Web Audio API's scheduling system to play audio in the future at a specific start time, on all devices.

So a central server relays messages from clients, telling them when to start and which sample position in the buffer to start from.

Interesting. Feels like this might still have some noticeable tens-of-millisends latency on Windows, where the default audio drivers still have high latency. The browser may intend to play the sound at time t, but when it calls Windows's API to play the sound I'm guessing it doesn't apply a negative time offset?

So it doesn't need to use the microphone? I guess from the "works across the ocean" comment and based on this description. I would have thought you would listen to the mic and sync based on surrounding audio somehow but it's good to know that it's not needed.

Yup no microphone. It's all clock sync

Thank you for the kind words! Yeah, I think it gets a lot more complicated once you start dealing with speaker hardware. It pretty much only works for the device's native speaker at the moment.

The instant you start having wireless speakers (eg. bluetooth) or any sort of significant delay between commanding playback and the actual sound coming out, the latency becomes audible.

For devices with mics, can you have them play a test chirp to measure the latency of Bluetooth or other laggy sound stack?

I primarily built this for group in-person listening, and that's what the spatial audio controls are for. But what is interesting is that since it only requires the browser, it works across the internet as well. You can guarantee that you and someone else are listening to the same thing even across an ocean.

Someone brought up the idea of an internet radio, which I thought was cool. If you could see a list of all the rooms people are in and tune it to exactly what they're jamming to.

> You can guarantee that you and someone else are listening to the same thing even across an ocean.

How can you guarantee that? NTP fails to guarantee that all clocks are synced inside a datacenter, let alone across an ocean (Did not read the code yet)

EDIT: The wording got me. "Guarantee" & "Perfect" in the post title, and "Millisecond-accurate synchronization" in the README. Cool project!

More, the speed of light puts a hard cap on how simultaneous you can be. Wolfram Alpha reckons New York to London is 19ms in a vacuum, more using fibre.

Going off on a tangent: Back in the days of Live Aid, they tried doing a transatlantic duet. Turns out it’s literally physically impossible because if A songs when they hear B, then B hears A at least 38ms too late, which is too much for the human body to handle and still make music.

It's a less hard problem than the duet. If the round-trip is 38ms, you can estimate that the one-way latency is 19ms. You tell the the other client to play the audio now, and you schedule it for 19ms in the future.

That's assuming standard OS and hardware and drivers can manage latency with that degree of precision, which I have serious doubts about.

In a duet, your partner needs to hear you now and you need to hear them now. With pre-recorded audio, you can buffer into the future.

yes but only after posting! it's very cool—i'm actually a little embarrassed to not have seen it before.

they're doing a smarter thing by doing streaming. i don't do any streaming right now.

the upside is that beatsync works in the browser. just a link means no setup is required.

thank you!