SAW TCP/IP and audio transmission

[Dingo]

Bob, would it be possible to add audio transmission over Ethernet from slaved machines? I don’t know about anyone else, but I would find that incredibly useful and amazing.

[cgrafx]

Bob, would it be possible to add audio transmission over Ethernet from slaved machines? I don***8217;t know about anyone else, but I would find that incredibly useful and amazing.

There isn't a good way to do this in a useful manner as just software.

To get low latency audio you need hardware (dedicated chips) to support it.

AVB (MOTU) or Dante or MADI could get you there now. A MOTU M64 connected to each SAW/SAC machine would get you 64 Chanels bi-directionally at around 2ms latency.

From MOTU's website

The M64 delivers an astonishing round trip latency (RTL) of 1.70 ms (milliseconds) on OS X and 1.71 ms on Windows over USB (at 96 kHz with a 32-sample host buffer)

[jmh]

Dingo,

In Voip applications a jitter buffer is usually employed to allow for delays in a packet switched network. The buffer can sometimes get large. This is why you get more double talk on discussions on cell phones or voip circuits. This delay causes us to start to reply when the the other end of the conversation continues to talk in a way that does not happen face to face or even on landlines. You have to draw the line somewhere and reject packets that don't arrive by a certain time threshold. That in turn forces you to use codecs that can partially recover from lost packets. Encoding and decoding introduce other delays. There are things like QOS intended to prioritize packets, but this entails careful network setup...

Anything that passes over the internet is prone to these issues. If you are doing point to point or on a network with limited traffic you would have a chance. However, ethernet cards and drivers are not particularly designed for audio.

The rml products are designed for accuracy in time so I would think anything in the above category is sort of pointless.

I have a Behringer xr18 which has a rj45 port that transmits according to a low latency audio spec. At the time I looked for a computer interface card to see if I could use this as my audio interface. It is possible such a thing may now exist and might be useful with SAC or some other application in the way you describe. Also, the behringer audio-over-cat5 had a much longer wire length than usb and possibly longer than than ethernet spans which is about 100 meters. I think they also have a mic-cable version too.

[Dave Labrecque]

Dingo,

In Voip applications a jitter buffer is usually employed to allow for delays in a packet switched network. The buffer can sometimes get large. This is why you get more double talk on discussions on cell phones or voip circuits. This delay causes us to start to reply when the the other end of the conversation continues to talk in a way that does not happen face to face or even on landlines. You have to draw the line somewhere and reject packets that don't arrive by a certain time threshold. That in turn forces you to use codecs that can partially recover from lost packets. Encoding and decoding introduce other delays. There are things like QOS intended to prioritize packets, but this entails careful network setup...

Anything that passes over the internet is prone to these issues. If you are doing point to point or on a network with limited traffic you would have a chance. However, ethernet cards and drivers are not particularly designed for audio.

The rml products are designed for accuracy in time so I would think anything in the above category is sort of pointless.

I have a Behringer xr18 which has a rj45 port that transmits according to a low latency audio spec. At the time I looked for a computer interface card to see if I could use this as my audio interface. It is possible such a thing may now exist and might be useful with SAC or some other application in the way you describe. Also, the behringer audio-over-cat5 had a much longer wire length than usb and possibly longer than than ethernet spans which is about 100 meters. I think they also have a mic-cable version too.

jmh -- doesn't the Behringer stuff use a proprietary protocol on the RJ45, intended for their digital snakes (not Ethernet per se)?

[cgrafx]

jmh -- doesn't the Behringer stuff use a proprietary protocol on the RJ45, intended for their digital snakes (not Ethernet per see)?

The interface on the X18 box is ULTRANET and I don't believe anybody makes an ULTRANET interface card.

[AntonZ]

Anything that passes over the internet is prone to these issues. If you are doing point to point or on a network with limited traffic you would have a chance. However, ethernet cards and drivers are not particularly designed for audio.

Yes you are right about that. However, I interpreted the original question not as a feature to route audio over the internet. The question as I read it is about "streaming" on a local network from one pc to another. It is still indeed a package switching network with inherent possibility of packet loss. UDP is the common approach, no guaranteed delivery of each and every packet. But if there is not too much traffic and the machines are on the same network (in other words no routing with multiple hops) then it should be feasible without too many hickups. There will however be no common workclock so there will occasionally be a small amount of error. Some other DAW's can do this, so I assume it can be done as long as you don't aim for bit perfect.

[jmh]

The question as I read it is about "streaming" on a local network from one pc to another. It is still indeed a package switching network with inherent possibility of packet loss. UDP is the common approach, no guaranteed delivery of each and every packet. But if there is not too much traffic and the machines are on the same network (in other words no routing with multiple hops) then it should be feasible without too many hickups. There will however be no common workclock so there will occasionally be a small amount of error. Some other DAW's can do this, so I assume it can be done as long as you don't aim for bit perfect.

I see your point. I would not be surprised if there was a virtual interface where a stream could appear as a device driver and could be accessed with any app.

In the linux world you could probably cobble this together with jack.

It might be of interest to look around on inet.

Going back to dingo's original question, I think I'm drifting from the point again. I think these social distanced performances we're hearing at this time is wrecking my mind and I can't pay attention anymore...

[Warren]

Bob, would it be possible to add audio transmission over Ethernet from slaved machines? I don’t know about anyone else, but I would find that incredibly useful and amazing.

I have been fiddling with ASIO LINK PRO a bit, seems to be tolerable on a LAN as long as demand is a few tracks.
It seems to have a lot of function. It cant be purchased and no support as the owner has passed but his son I believe, offers it for FREE! and offers a patch.

https://give.academy/downloads/2018/03/03/ODeusASIOLinkPro/

worth a try for some needs and the price ain't bad.
Have fun!

Warren

[Dingo]

Thanks for all of the replies! Very helpful.