Dream Scenario: AV over IP in my Homestudio - the linux way

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Dream Scenario: AV over IP in my Homestudio - the linux way

Post by Corfromleuven »

I need some help with the set up of a home studio I want to tinker with. There is some interesting stuff out there in terms of Audio over IP nowadays. You have mikes and active speakers, but also Class D amps that are POE capable and they can decode any fileformat you care to listen to. I have ethernet CAT cables running everywhere in the house. In theory I could like to use them differently in the future. To provice POE++ for instance to all my devices, by using an active switch. I don’t need very heavy amps and loads of power. This is hobby stuff I want to install in the one spare room in the house to play with and do personal audio projects.

Before I start dreaming, what I would like to know is: in how far can you use current audio hardware (speakers but also microphones) that are DANTE/AES67 compatible with a standard linux server/player-recorder setup?

I have two questions that bother me with the setup I have in mind:

Let’s assume I buy some a bunch of AES67/DANTE speakers and mics. I will be able to connect them to a POE cable connected to an active switch. All devices would then receive their own IP address from startup.

To record/play back the recorded files I would use a Mediaserver/Player setup. A player device any basic x86 (even RPi5 with POE theoretically) would receive a stream from the mediaserver over the CAT cable via the active switch as well as power if I choose to add POE into the mix.

Now, before I buy any hardware, I would like to know how well this could work with a relative standard linux setup like Ubuntu Studio? If I understand more or less what pipewire and wireplumber can do, a PC with these installed should be able to detect the AES67 compatible devices (speakers or microphones) in any IP network. They would normally show up in the pulse volume controls so I could see them and select them from there in order to use them in Ardour for instance.

I assume here that, in principle, the one PC responsible for the capture and playback should be able to be the master(clock) and serve all AES67 devices properly.

My first question is: how automatically would this work and continue to work?

Would the player detect devices at each startup, almost automatically? Every time I switch the speakers or mics and the players on again? This is no theoretical scenario. I am a very energy savvy person and, I would, in fact, prefer switching the entire setup off every time I consider a project done, or go on holiday for a couple of weeks, in order to save energy.

My second question is: how does this system “knows” which speaker or mike is which (Front, Back, Left, Right)? Are interfaces like pulse or ardour (or qpwgraph) capable of holding the choices you make in the hardware install phase, persistent after each shutdown/startup of the system?

Thank you for any help.

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