https://news.stanford.edu/2020/09/18/jacktrip-software-allow...
Its also a peer2peer system, and not easy to set up correctly.
Some tests that focus on interconnection issues (ISP to another ISP):
Continuous
To Azure: https://azurespeedtest.azurewebsites.net/
To AWS: https://awsspeedtest.xvf.dk/
One-time
To Cloudflare: https://speed.cloudflare.com/
To Netflix (maybe misleading due to direct interconnectivity in some ISPs): https://fast.com/
Even just the difference between local loopback (instrument -> headphones) vs instrument -> USB -> PC -> USB -> headphones, feels like playing with someone continually dragging the tempo down.
I'm sure I could relearn and adapt to >10ms, but this is a far cry from a live experience of playing with others, and I am definitely not alone in my cohort (after trying several low latency remote meetup configurations).
Many of us in classical music are doing this all the time. So it's definitely doable. Especially if you have a conductor! Indeed, that's one of the reasons conductors are useful.
There are some good instructions here for different systems.
The old analog POTS phone system wasn't like this. It had that feeling of almost being in the same room, no latency, true birectional conversation.
I was wondering if there's anything that tries to get close to this old POTS level of quality, at the cost of higher bandwidth and processing power.
Can some of the software mentioned in this thread achieve this?
To make it work well you Amy have to turn off lots of helpful processing that modern systems do.
I mostly agree with this, but POTS starting going digital surprisingly early, before VoIP ever became a thing. And of course digitization back then was typically chopping a huge amount of the high frequencies, making things sound...well, like the phone.
Most people making calls today have no conception of how good telephones used to be.
I'm a musician and I'm incredibly confused by this point. Music instruction cannot possibly be effective without a visual component, regardless of whether it's a solo instruction or a group. An A/V method with a bit of latency is almost always going to be superior to audio-only.
Even just touching strings in different ways can produce wildly different results; think playing a harmonic note on a bass guitar, which is how you get such a low-sounding instrument to play a high-pitched note. How you blow air through your horn can greatly change the timbre and mood of your music.
There are a lot more examples, and they are all difficult to correct if you aren't able to see what the musican is doing.
I suppose blind people will never be successful with music.
Blind people are perfectly capable of learning to play music. Blind people and non-blind people alike can learn on their own, or seek out instruction. It is the latter that we are discussing here.
I’ve been a hobbyist musician since leaving school 20 years ago. I play piano, guitar, bass, mandolin, cello, drums, etc. All of the typical video platforms fall short of having a low enough latency to achieve “remote orchestra”.
There was one product/service that I tried that got closer than anyone, JamKazam (https://jamkazam.com/) and there were a few audio only options back mid-2000s that had close to 20-50ms which was better.
No, it will not. :) You can get to 5ms above the network delay, even using compressed audio (opus), though some sound devices may be picky.
Network delay can be just a few milliseconds even across a city, assuming that no latency murdering devices (wifi or nics in interrupt mitigation mode) are on the path.
This means that you can have lower audio delay from a compressed audio conference crossing your state than is achieved from speed-of-sound delays from a performer sitting on the other side of a moderate sized room!
The future is already here -- it's just not very evenly distributed. (yet)
If you limit your collaboration radius in that way, then sure (though it still depends on the router technology the packets encounter along the way). But if you want to go beyond that, the numbers get bigger. For transcontinental stuff, much much longer.
But thanks for correcting the basic thrust.
This is the same one way delay as sound in air across 12 meters, which would be on the large size for an orchestra pit.
It does get larger as you go further, but live performance with people on the other side of a large continent is completely realistic and has been done. :)
Sure, expecting to go between NYC and someone in china behind the GFW is probably asking too much for a seamless experience. But many people mostly want to work with people in the same country as them ... and for that, with sufficient technology, latency need not be an issue.
In case you're not aware, I'm the original author of JACK, around which JackTrip is built, which is the most likely and reliable tool for such a collaboration. I certainly know people who've done this, and I regret that I didn't realize back in February how useful it would have been to make something like JackTrip into a much easier-to-use tool for computer-naive folks.
37ms cross-country in the USA is optimistic, but certainly possible. US->Europe is not so great.
When the pandemic hit I thought about putting out some pointers to easier to use low latency streaming resources... but... that would require overcoming pandemic lockdown funk. And honestly it's still kind of a technical rats nest to get everything working. It's not exactly musician friendly to need instructions like "next you need to make sure your nic doesn't enable interrupt mitigation when there are more than 100 packets per second...". :)
[I love Jack BTW.]
One of the 2 netjack implementations uses Opus - great stuff.
Absolutely right that the good stuff (i.e. not zoom) requires way too much setup, and even then (as you note) you're not guaranteed reliable function because you don't control all the hops. Soundjack gets the user-side fairly good, but it's still not quite what I think would have really taken off during the pandemic - probably needs a mobile app for that.
And hey Paul, long time no chat!
It sounds do-able. It sounds like we should already be there. Looking around though you’ll see we aren’t because of technical issues described that are systemic of the infrastructure used.
He's been teaching music remotely since August, and one of the first days he demonstrated why they have to be on mute on Zoom by having them all sing happy birthday to one of the students. It was utter chaos. I worked with him to try to find a solution, we tried various tools, but found nothing that worked across cable, DSL, WISP, and cellular connections.
Some tools could sync after the fact so the recording sounded good (if each musician keeps the same tempo), but the whole point is that the kids needed to be able to hear and follow each other, and just one student at 500ms latency totally ruins the tech.
So he teaches music to second graders by zoom, and they're all on mute unless he unmutes them one by one.
Yes, it will as soon as you are 1000km away. Light speed creates a lot of unresolvable latency. How can you play live between London and Sydney? You need a different musical understanding.
The real problem is computer audio isn’t designed for low latency. Generally, as long as video and audio are reasonably synchronized nobody notices, but latency as rarely a consideration on it’s own.
Even with the state of the art equipment, transcontinental music making will always exclude the kinds of music that most of us play while in the same room together.
AWS latency isn’t great between regions. Looking on some other sites you can see London to NYC is 70ms, but that’s also 5,500km which is far behind the 1,000km mentioned. Ex: https://wondernetwork.com/pings/London
70ms /5.5 = 12.7 ms seconds round trip. The speed of sound is only 1,125 ft/s, and half that for round trips so 1,000km over fiber is the equivalent of ~7.14 feet apart in the same room plus whatever latency your personal computer and software adds.
Well, I could probably get some form of POTS either analog or ISDN/T1
If you use sonic, at least in some areas they have a setting where you can turn down the level of forward error correction and get lower latency in exchange for somewhat higher packet loss.
Sure. If you constraint it a little more, you can also just play to the persons in the next room directly (no streaming required).
For country level or worse global collaboration latency will always be a problem however...
If your conferencing is 'slowing down' and 'speeding up' then it's simply not a realtime communications medium.
Even with fiber all ISPs I have ever used at home have latency murdering devices between my modem and their connection to the Internet. And I have to go through these hops even if I connect to an IP on the same IP.
Best I ever got was 12ms. Once you connect your laptop to the Internet in a data center you will instantly know what I mean.
Got a Unifi AC LR and it only adds about 4 to t 6ms over Ethernet.
This is my understanding as I’m still learning. Fascinating this stuff works at all.
Unless everyone just jam to a click track instead of each other, it's very hard to play together without seeing the body language.
In other words, they don't want a real time music streaming app, they want a music teaching app. Duh!
This makes sense if you are trying to teach, you need to see the techniques not just hear them.
Which may be tied to its low latency... I assume there are algorithmic reasons why cancellation needs latency.
That and between the audio setup wizard and its unorthodox auth system, the onboarding process is brutal for non-technical users.