I solved this by decompressing the whole file into memory as PCM.

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.

Hopefully wifi8 has something PTP built-in. I hear there's some vague hope that better timing info is one of the core pieces, so maybe maybe!

I'm super jazzed seeing AES67 emerge.. although it not working great over wifi for lack of proper timing info hurts. Very understandable for professional gear, but there's nothing I love more than seeing professional, prosumer and consumer gear blend together!

PipeWire already has pretty decent support! There's a tracker where people report on with their hardware experiences trying it. Some really really interesting hardware shows up here (and elsewhere on the gitlab): https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/32...

Well, AES67 isn't a great secret... It's just RTP with PTPv2, with some predefined codec, sample rate, and fpp options.

Absolutely! Silent disco still requires impractically expensive rental hardware to work well as far as I know. A lot of them run off FM radio, since it's the simplest way to go, but nobody owns portable radios anymore.

An OSS app with the ability to sync everyone up over mobile or wifi, on Android or iOS with BYO headphones, would be incredible. This should be a thing :)

I believe the syncing won't work when playing with a bluetooth device

"Their own source" looks like they are bringing their own files or (more probably) their Spotify or YouTube. It happens all the time on public transport. Or did you mean bringing their own music and taking turns at sharing it with the other people around? That might be against the terms of service of some services.

For those wondering: The rule thumb here is that light travels at one foot per nanosecond. 300 ns =0,3 μsec. Electricity is a bit slower but the same order of magnitude.

Oh, thanks for correcting me! Now that you mention it, I'm confused by a memory I have. Wired speakers seem to be less common these days but I remember being told about two decades ago that the "proper" way to install speakers was to run out equal lengths of speaker cable (basically just jacketed copper, afaik) to different speakers even if they weren't equidistant in a room. (This was advice for home installation, not stadium-sized installations.)

Do you suppose there exists some other reason for that, like maybe matching impedance on each cable, or is this likely one of those superstitions that audiophiles fall prey to?

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

Awesome project!

If you support mic input, you can allow the user to select a device as the "nexus" with mic recording on. Then you tell each device in your setup to "chirp" at the same exact time, but at different frequencies. Then you can derive the individual device's "local delay" and compensate.

This allows you to tune the surround setup to full accuracy for a given point in space, and it will take care of ring buffer differences, wireless transfers of non-teathered speakers, etc.

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.

Haha yeah guarantee is a strong word. I just mean that it’s good enough to not be noticeable (even within the same physical room)

Oh, that's a nice approximation! Similar to Grace Hopper's famous demo of a six inch wire being about how far electrical signals travel in a nanosecond.

Wow it's insane how slow that is!

But also, I don't really have an intuition for why the speed of travel is relevant here?

It's funny that I have a natural intuition that sound is slow over long distances, yet 1 ft/s still feels astonishingly slow. And yet while I know light travels 1ft/ns, it's still astonishing that it takes 30ms to travel from London to Sydney at that speed.

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

This is kind of where my attempt at this idea during lockdown died... Copyright law

The challenge here is not technical but legal, good luck.

Made an update so it should be good on most!

Yea could be, this was using their .gg hosted version, I'll try locally

I was eagerly expecting such practical breakdown of the Vision. Thanks for the numbers.

But, the scenario is of a Open-Source solution for a Student living in a Hostel/ Home.

The Vision is tilted towards Hostel Dorms where you won't bother a Home Theater, but where every friend has Apple device( any robust device with good onboard audio).

My Friend himself has a MBPRO and 2 iPhones, Totalling to 6 Apple device( Audio Sinks).

Here are my server and client configs needed in case someone comes across this from google. It sets up sinks and sources, so you can just mute it, but it would just play automatic from logon:

Needed on Server and Clients is an override to a) fix my domain users having the same cookie if its stored in default location and b) make sure the server only starts when the network is REALLY up - the normal network online is a system service only and thus you cannot check for it in a users service. In my case the server runs under a domain users profile.

~/.config/systemd/user/pipewire-pulse.service.d/override.conf

  [Unit]
  After=user-network-wait.service avahi-daemon.service

  [Service]
  # this changes the location of the cookie because i use roaming homes for domain clients and each machine would have the same cookie
  ExecStartPre=/bin/bash -c 'systemctl --user set-environment PULSE_COOKIE=/run/user/$(id -u)/pulse/cookie'

~/.config/systemd/user/user-network-wait.service

  [Unit]
  Description=Wait for Network Connectivity

  [Service]
  Type=oneshot

  # This pings your LAN router and creates a network-online file in /run to pick up
  ExecStart=/bin/bash -c '[ -f /run/user/$(id -u)/network-online ] || (until ping -c1 10.126.0.1 >/dev/null 2>&1; do sleep 1; done; touch /run/user/$(id -u)/network-online)'
 
  [Install]
  WantedBy=default.target

Server Pulseaudio:

Not needed but very useful:

/etc/pipewire/pipewire-pulse.conf.d/50-networkparty.conf

  context.exec = [
      { path = "pactl" args = "load-module module-native-protocol-tcp auth-anonymous=yes listen=10.126.1.1 auth-ip-acl=127.0.0.1;10.126.0.0/16" }
  ]

# needed. Note how to to make sure s16le is used across all devices to keep conversion to a minimum and how to name the sink somewhat sane

/etc/pipewire/pipewire-pulse.conf.d/70-rtp-sender-sink.conf

  context.exec = [
    { path = "pactl"        args = "load-module module-null-sink sink_name=rtp_sender_sink format=s16le channels=2 rate=48000 sink_properties='device.description=\"RTP Sender Sink\"'" }

]

/etc/pipewire/pipewire-pulse.conf.d/71-rtp-sender-23912611.conf

  context.exec = [
    { path = "pactl" args = "load-module module-rtp-send source=rtp_sender_sink.monitor source_ip=10.126.1.1 destination_ip=239.126.1.1 port=5004 inhibit_auto_suspend=always" }
  ]

You can play to the sink f.e. in mpd with:

  audio_output {
    type "pulse"
    name "RTP Sender Sink Pulse"
    sink "rtp_sender_sink"
  }

Client Pulseaudio:

/etc/pipewire/pipewire-pulse.conf.d/71-rtp-receiver.conf

  context.exec = [
      { path = "pactl"        args = "load-module module-rtp-recv sink=combine_sink sap_address=239.126.1.1 latency_msec=64.3750" }
  ]

you can play with the latency_msec, journalctl will tell you the lowest fragment if you just put 0 or 1ms here. It needs to be a multiple of that minimum, just experiment. Im fine with this even though 12ms would also work in my lan, but its more stable across the wifi bridge.

The sap_address on the client may work to select the right multicast address even though its actually for the SAP announcements but don't count on that, i have not tested multiple streams so far and would not use "magic" solutions like SAP on the server (and they didn't work in my case and seem pipewire-only). Right now the client seems to pick the right stream - experiment ;)

The sink in my case is a module-combo, just check with pactl list sinks which sink you want the stream to play on. Note that this is not some application you can dynamically assign to other sinks!!

For LAN, if you run openwrt just enable igmp_snooping and multicast_querier on the softwarebridge (Luci --> Network --> Interfaces --> Tab Devices) and maybe Multi to Uni in your wifi advanced settings. I dont use this though as my wifi is another vlan or WDS-bridged so i stay out of these problems mostly.

There are more advanced settings possible with openwrt, including having working igmp_snooping on the hardware switch, if you are interested frequent my documentation (german) on Krei.se as i will write a guide for this sometime lol (or just ask me by DM). Its possible to run this ms-exact with clean network in any case, there is no need to install extra software or clog unused ports with multicast-traffic. If you are perfect about this the music will flow like water through your LAN only where its needed.

Haha, well you have to set up multicast in your LAN like a good plumber, then fight context.exec and context.modules - also "just works" means in my case setting up a systemd After=network-watchdog in pulseaudios override that checks whether network is up in userspace.

But Poettering Software (systemd/pulseaudio) is quite composable so even though there is a learning curve the alternative are 20k config file monoliths.

Still this even turns rooted LG TVs and cheap raspi picos into sinks.

The only latency i now have is the bass traveling slower than the trebles lol