- #Ableton live 9.1 change the overall latency setting full
- #Ableton live 9.1 change the overall latency setting software
- #Ableton live 9.1 change the overall latency setting code
I’m building a desktop utility for guitarists using clojurescript and Tauri. I’ve been formulating a similar idea for a while now.
#Ableton live 9.1 change the overall latency setting full
There are artists who insist on using a full analog chain for monitoring for that reason, and refuse to use digital mixers or digital wireless systems that can add several ms of latency. I've even heard of people successfully using protocols like Dante (professional audio over IP intended for LAN) over gig fiber lines as well.įrom my experience, a couple milliseconds of latency is fine for keyboard or guitar processing, but anything more than that starts to mess me up. So I suppose this tool would work well fine over fiber assuming everyone is in the same city. With fiber, you can hit the exchange point in 1-2ms. I've actually tried this experiment with some of my nearby friends - pinging each other's public IPs on DSL or cable takes around 20ms. I live in a major city and it takes around 10ms to hit the internet exchange point (using Ethernet the LAN latency is neglegible) and thus I get a minimum of 20ms or so just connecting to a server hosted by someone in the same neighborhood.
Yeah, the hop to the internet exchange point is the main issue with DSL or cable. I expect it will still be useful in internet rich areas like city to city. I like the idea, but feel like the internet isn't there yet for the majority of users, and latency hasn't exactly been improving at a great pace.
#Ableton live 9.1 change the overall latency setting software
If anyone has tried playing a processing heavy software synth on a pc in real-time you will have experienced how unplayable it is as soon as latency goes beyond 10-20ms - you can't play music if there is noticeable delay between your fingers and ears, and we are much more sensitive to sound latency, it would be the same problem trying to play in time with each other. which most people don't have, although I realise a lot of the tech crowd is unaware of this (most of the worlds user end points are some kind of DSL or cell network with a 20-40ms minimum). It's good that latency is considered to be so critical, but for the same reason I'm sceptical that this would work well for the majority beyond quite local ranges with very good internet connections i.e some kind of fiber. You may be able to improve on this but disparaging it with silly criticisms isn't going to help you there are thousands maybe millions of satisfied Jamulus users around the world. "The only clear benefit to it."? As I suggested, a significant benefit of the Jamulus model is that any of the clients can be "thin" most of the computation is done on the server. In the multiple-ISP scenario I had to deal with, a server "midway between all the users" would have been useless, as would any kind of server-less topology. If none are suitable or one wants a private server, one can set up a server anywhere that provides good ping times to all the intended clients. When one tries to connect in Jamulus, dozens of public servers are suggested with their ping times listed. Did you read my post? The server is in Montreal all the clients are 200 miles west of Montreal in Kingston. "need to find a server midway between all the users" Nonsense. Interesting to hear about other's solutions though! 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. Local recording of your end with some shared sync markers, that you could manually or automatically sync up post-hoc
We had trouble setting up our instrument, but also adding a mic for chatting. An easier way to join with multiple audio inputs. Auto register new audio devices when you plug them in/edit in settings.
#Ableton live 9.1 change the overall latency setting code
Join room by name (we were confused that there was a different code than the name we chose) 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.
Once we had that tuned it worked really well. 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. I had a go at jamming with my brother yesterday. Interested to see how it fares in a 3+ participant setting. 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. The system seems to have the settings right for realtime collab. The packet capture didn't seem to be RTP, are you using libwebrtc under the hood? Have to say this is the best version of something like this that I have tried.