what is delta encoded ? ADPCM ?

Could the algo prevent clicks in the case of packet loss in low latency streaming ?

woah what is this fastpast magic ?!

Honestly I was a bit blown away when I got it working the first time ! It kind of mental that I can capture audio and compress it - send it through all these layers and machines - and receive it 1000km away 20ms later. All on consumer grade internet.

I used the standard 2.5ms. I know you can go lower if you want, but then you need a higher bitrate as it's "custom".

I turned off FEC as it adds latency.

Jitter buffer right now is just user controlled. A bit lame, - I should make it automatic, but need to get the right heuristic. With a LAN connection, the buffer can be as low as one or two packets.

I don't use any packet redundancy either and there's no ARQ as if you have to ask for a retransmit you've already lost the war!

How about you ?

Yeah, awesome. Have to say this is the best version of something like this that I have tried. The packet capture didn't seem to be RTP, are you using libwebrtc under the hood?

The system seems to have the settings right for realtime collab. In fact I found it much better for chatting too if you can guarantee they have a headset and good mic - I hate current-gen AEC algorithms. Interested to see how it fares in a 3+ participant setting.

I had a go at jamming with my brother yesterday. We had to adjust down the delay to 2ms (I think default was 5ms) in order to counteract the "lagging" effect you get when you lock into the remote beat. Once we had that tuned it worked really well. We would occasionally suffer burst losses, but you can play through it and it's still synced afterward which is the best you can do with an unreliable network.

Some feature requests: - Join room by name (we were confused that there was a different code than the name we chose) - Auto register new audio devices when you plug them in/edit in settings. - An easier way to join with multiple audio inputs. We had trouble setting up our instrument, but also adding a mic for chatting. - Local recording of your end with some shared sync markers, that you could manually or automatically sync up post-hoc

The use case I use WebRTC for is slightly different, where quality >> latency , so we have it tuned to the other end and run the full gambit of FEC, ARQ, packet redundancy etc. Interesting to hear about other's solutions though!

Hey thanks.

You can also send the backing track ahead of time a little so that you both hear it at the same time!

Do you mean that you want to mix the inputs from multiple devices ? This is possible - but you do get a little bit of extra latency of course.

It's supposed to refresh the audio devices when you "mouseenter" the select box, so perhaps there's a big there.

Multitrack recording with a bigger buffer would be great.

Good you found the buffer settings. I'm surprised that you found a significant difference though between 2ms and 5ms?

no retransmits - Opus handles a degree of packet loss pretty well. Not using FEC as it adds extra latency.

Yes JUCE is da bomb

Thanks!

I haven't ruled out opensourcing, but honestly I already have limited time and in my experience open source takes _more_ time commitment (I get that you will get free help eventually).

I'm making a VST plugin to stream output from a DAW.

Problem with Tauri is that you have to support the native browser, rather than just chrome, so it's more work to build and maintain.

Good luck with your project!

J

I would like to make some money with it to continue the support, but it's not clear yet the best way forward. Currently it's donationware, but other models could be a subscription, or some sort of premium/freemium model.

Really I just want to see how people use it and figure it out from there.

Thanks !