The real-time low latency multi channel audio streaming needed for musicians is awfully similar to the real time low latency multi channel audio streaming required for telephony.

Yet somehow the two industries have pretty much entirely different tech stacks and don't seem to talk to one another.

This is very much not true.

Telephony is significantly less latency sensitive than real time audio processing, it’s also significantly less taxing since you’re dealing with a single channel.

The level of compression and audio resolution required are significantly different too. You can tune codecs for voice specifically, but you don’t want compression when recording audio and can’t bias towards specific inputs.

They’re only similar in that they handle audio. But that’s like saying the needs of a unicycle and the needs of an F1 car are inherently the same because they have wheels.

Most telephony I've experienced has latency measured in seconds (if you ever call your friend or spouse sitting next to you it becomes very obvious :) vs audio recording and processing which is measured in milliseconds.

Additionally, from what little I'm aware of, telephony is heavily optimized for particular frequencies of human voice and then heavily compressed within that. As well, any single telephony stream is basically a single channel. A song may have dozen of channels, at high resolution, full spectrum, all sorts of computationally demanding effects and processing, and still need latency and sync measured on milliseconds.

So... Kind of the opposite of each other,while both being about processing sound :-).

I feel like equating telephony and music production is like saying writing firmware and a HTTP/JSON backend for a website is the same. True, both are programming I suppose, but vastly different requirements, assumptions and environments.

This is a very interesting thought. I'm not super experienced with low level audio and basically completely ignorant of telephony.

I feel like most people doing audio in music are not working at the low level. Even if they are creating their own plugins, they are probably not integrating with the audio interface. The point of JACK or Pipewire is to basically abstract all of that away so people can focus on the instrument.

The latency in music is a much, much bigger issue than in voice, so any latency spike would render network audio completely unusable. I know Zoom has a "real time audio for musicians" feature, but outside of a few Zoom demos during lockdown, I'm not sure anybody uses this.

Pipewire supports audio channels over network, but again I'm not entirely sure what this is for. Certainly it's useful for streaming music from device A to device B, but I'm not sure anybody uses it in a production setting.

I could see something like a "live coding symphony", where people have their own livecoding setups and the audio is generated on a central server. This is not too different than what, say, Animal Collective did. But while live coding is a beautiful medium on its own, it does lack the muscle memory and tactile feedback you get from playing an instrument.

I would love to see, as you said, these fields collaborate, but these, to me, are the immediate blockers which make it less practical.

irony amplified by the nature of the tech stacks xD surely they can figure out some channel to communicate over clearly haha