Integrate audio over IP protocols in the media kit

A lot is going in audio over ip (aoip); things that traditionally belong in the analog domain are now moving to IP networks in all kinds of venues and studios alike, greatly reducing costs and increasing flexibility. What this also means is that the traditional sound card, the interface between computer and other analog or digital audio gear, is dying out and is going to be replaced by networked interfaces. The computer is already IP ready!

Even though there are two open standards for layer 3 aoip (aes67 and ravenna) no os supports them out of the box. There are proprietary drivers available for linux, os X and windows but no open implementations or support in the os. Apple, claiming os X to be THE os for audio, really should have implemented aes67 and ravenna the day they went public… But they didn’t.

There is a free wdm driver for Ravenna support from Alcnetworx but I don’t think the driver itself is open. Also, wdm is not, to my knowledge, usable for low-latency audio.

Implementing aoip standards in Haiku could attract a lot of attention from experimental and professional audio and would also solve the problem of bringing support for professional interfaces to the platform; something that has plagued linux audio since always. RME has long been the only serious choice available for linux.

This is definitely the future for audio and It could be cool for Haiku to be on the train from the start.


Ravenna (superset of, but compatible with, aes67):


Here are some Ravenna/Aes67 interfaces. Supporting aoip standards means also supporting these very serious devices.

Merging HORUS

Merging HAPI

Axia Xnodes


Directout Montone.42

The Dante protocol from Audinate interoperates with AES67, so supporting the AES standard would also bring support for a whole bunch of devices, including Yamaha and Allen & heath consoles and focusrite IO.

Just sayin’

Here is the start of an open source implementation of AES67… No code yet, though.

I’ve suggested implementing an AoIP driver to Tune Tracker Systems before, due to their issues in obtaining suitable audio cards for broadcast environments. Its not a particularly easy thing to do so having a commercial impetus to do so might be a help.

More than anything else I don’t know if the multicast support in the network kit would be up to it as it stands.

I work in a major broadcaster who has removed all analogue audio chains from our main radio stations at this stage, moving entirely to Axia Livewire and it really is a game-changer in integrating computers to the audio chain. Any analogue source used goes directly in to a node and all the desks, routing etc is 100% Axia.

That’s awesome! I wish I could do that in my recording studio. Considering all our old investments in Pci audio interfaces, AD/DA and walls full of copper it’s hard to convince people to throw it all away for something like a Horus, though.

So, what did did the people over at tune tracker systems think of your suggestion?

Is it imaginable that some day the multicast support would be improved?

By the way! Are you using standard integrated ethernet ports or some kind of specialized hardware for your computers?

We moved studios so it was a perfect excuse - did the buildout entirely new in the new ones and linked back over ISDN on to the old desks until the playout was changed.

Dane seemed interested but I don’t know if anything more will come with it. Multicast support will likely be improved over time.

We use conventional gigabit ethernet cards, usually the original onboard with a generic discrete (or occasionally USB in some cases of urgency) network card for normal networking; and Cisco gigabit switches dedicated for the audio network