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This is incredible. I've been wanting a piece of software to stream high quality, low latency mac audio to my iphone, so I can use my headphones wirelessly while working. I'll have to play with it more, but I was set up in mere minutes and it works at extremely low latency.

The fact that this does bidirectional audio is icing on the cake; can't wait to share this with some quarantined bands.

Magic, well done.

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15 years ago I played in a band with some friends.

We now all love in different cities. Due to the pandemic and lockdowns we ended up reconnecting and despite the distance we are jamming and working on songs like we never left the drummers basement.

Strongly recommend jamulus.io. It is fast enough to actually jam on music with other players.

How? The network itself should be 15ms, which is the perceptible edge or reverb. Genuinely heavily interested.

The answer is pretty obvious: by not caring if it is under 500ms delay.

unless you get a Quantum A.I. that can predict seconds into the future... mine just said musk will start a Q.A.I. company with some shady zero-equity A-round disguised as pre-sale. Oh wait that's not my QuantumAI, it's just my tweeter tab.

500 ms?! At 500 ms I can assure you you'll have trouble with normal conversations. You're nowhere near collaborative music playing.

everything on that comment was exaggerated for comedy. But the truth is still the same, you have to accept the network latency. period.

anyone promising faster than light communication could very well be claiming a patent to a perpetual motion machine.

Yes. Transatlantic latency is <100 ms. People in this thread have said that 50 ms is doable for making music. Why are you derailing the conversation?

It is perceptible, but doesn't matter that much in practice. If you're able to achieve the overall latency of around 30ms, you'll be able to play just fine - your mind will quickly adjust to the latency, you just need to listen to your delayed output from the server and disable any local monitoring so you stay synchronized with others.

As a musician I kind of strongly disagree with that. I was never able to adjust to latency above ~8ms. Some people might. I can't. I spent as much effort as necessary to ensure that my setup has low latency.

Of course if you're playing with quantization on, or sequencers, it's not as much of a problem

That always struck me as the solution: pump a click centrally to everyone and then everyone synchronizes to it. You might not be able to monitor but you've "jammed" and actually created music.

That's pretty much what Jamulus does: sound is mixed server-side, so you have a single point of reference (can be click track, can be drummer, can be sequencer...). You can even get a multi-track recording saved on the server so you can then adjust timings and give it a proper mix afterwards.

15 ms is the same as people standing 15 ms * 300 m/s = 4.5 m apart.

You can make music together at this distance (even though it's perceptible).

Hmm… Wonder whether it would improve things to create a virtual environment with point sources actually that distance apart, and model the reverb / echoing you'd get from that, so your brain "intuitively" understands the delay.

The most important thing when you do this is that you don't listen to your local audio signal, you only listen to the mixed signal coming from the server. That puts you in time with everyone else, more or less.

Roundtrip latency was under 30ms, which you definitely feel but your brain can compensate for.

The funny effect overall though is that because you're delayed and everyone else is delayed, you end up sort of waiting for the other player, and they end up waiting for you, so instead of the classic band "everyone speeds up" it's like everyone progressively slows down as the song progresses.

We haven't tried with our drummer actually playing drums, he's just been recording against other parts we provide. I don't think it'll work very well because his acoustic kit will be too loud and he'll hear that as well as the delayed drum sound, which for him will feel like a stuttery mess

> don't listen to your local audio signal, you only listen to the mixed signal coming from the server ... I don't think it'll work very well because his acoustic kit will be too loud

Yes, that sounds right. Whether this sort of approach works depends a lot on whether your instrument is naturally quiet enough that you can just focus on what you hear from the server. People are actually surprisingly good at adjusting to instruments that are "slow to sound"; just like you can bounce a basketball on the beat with a little practice, even though it takes many milliseconds for the ball to hit the ground after leaving your hand, you can learn to play keyboard etc with a similar delay.

Yep!

We had a zoom video going at the same time so we could look at each other. What tripped my brain up the most was if I looked for rhythm cues from the video while playing, even my own video would not be in sync and I would lose the zone where I could play in delayed time and have to restart

This use case can be done much simpler; this is something I've been doing with PulseAudio on my phone for years.

Could you elaborate on your setup an usecase?

I just enable zeroconf discovery and allow network access in PulseAudio on my laptop and my phone, so both devices automatically see each other as a regular audio device. You can do that from UI via `paprefs`.

https://news.ycombinator.com/item?id=8863

Have to link to that because I chuckled at your post.

Is that relevant though? ;) I don't think "using mobile phone as a wireless headset" is a primary use case for low-latency P2P apps and I certainly wouldn't propose using PulseAudio instead of such apps for, well, low-latency P2P communications.

Though PulseAudio doesn't really minimize latency ...

It seems on par with a Bluetooth headset to me. Of course you have to use it over LAN and not over the Internet, but I don't think that's an issue for this particular use-case.

I use Soundwire for the same purpose.

Out of curiosity, what's your use case?

I have a nice pair of wired headphones and I listen to a lot of lossless audio. I like being able to get up from my desk to fill up my water glass or grab a snack without breaking the flow of music. Bluetooth isn't ideal due to the quality loss and other factors. I tried some airplay based solutions but the 2 second delay was too much. I'm seeing anywhere from 25ms-200ms latency here and it's working flawlessly.

I could just listen to music on my phone directly, but it becomes such a pain because watching a YouTube video or anything with audio on my computer completely breaks my flow.

Granted this is a completely niche persnickety problem, but what is this site if not for solving niche persnickety problems.

My friend who happens to be an acclaimed tubist just noted that he could use this for online music lessons. So there's earnestly a wide variety of use cases here.

For what it's worth, I have the exact same use case

Have you given snapcast a look? https://github.com/badaix/snapcast/

> Granted this is a completely niche persnickety problem

I'm in the exact same boat - completely resonate with your desire to avoid breaking flow - and I doubt I would have thought of this solution if it weren't for your post above.

Thanks.

I can't figure out how latency matters when you're listening to prerecorded music. You hear the whole song from start to finish without interruption, yes? Why does it matter if that whole uninterrupted song is heard 2 seconds later than when it was sent from the streaming device? Oh... you're watching the music video at the same time?

Perhaps it is a usability issue. Low latency makes using the software much more pleasant. I once tried streaming lossless audio to my phone from an mpd running on my home server. Pressing pause and having to wait two seconds for the audio to actually pause killed it for me.

That’s a fair question, but I do a lot more than just listen to music. I jump on a call, I watch a YouTube video, I put on a basketball game or even the news. Those are all latency sensitive.

Why does latency matter for any of these except the call where you're interacting with the stream?

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