Remotepc forums11/29/2023 I hope you're recording and storing the audio on the server. It doesn't really matter, it's just to make the distinction from streaming. That saves bandwidth, but costs CPU on the server. It's a protocol and the video from the server will be encoded - even vectorized in RDP. I want to mix in 96k, or even 192k if possible, because I get a lot of aliasing, but I only need to 'hear' 48k, and rather than making the host do extra computational work down-sampling the master out, if there's enough bandwidth for 1080p video and, 24bit/192k audio, without noticeable latency, then my plan is gold.Ī VLC or RDP connection is nothing like you describe. There are still others, like Telos, who already have partial compatibility with AES67:īeing open doesn't exclude being compatible with closed standards.Īnd of course there are some smaller projects, like the EBU's ACIP:īasically I'm only sending keystrokes *to* the server, and *receiving* 1920x1080 video and 24bit/48k audio to my laptop. Dante is closed, so others wanting to implement Dante, will have to pay Audinate for licensing. Then there is Ravenna, which is a large part of the new protocol:Īnd among the first compatible networks, is one from Wheatstone, a broadcast supplier:Īnd the new standard is only possible because Ravenna is a mostly open protocol. There is finally some agreement between the industry, the EBU, the AES and a couple of others about a joint, compatible standard, called AES67, that would make machines like the X32 compatible with gear from other manufacturers. There's a lot of stuff to read on Ethernet AVB. When controlling a Windows box, I would go for one of the MS Remote Desktop clones. It wasn't with REAPER, tho.ĪRD (Apple Remote Desktop) was the remote control app of choice. I used a similar system years ago, with an old Powerbook G4 to record 8 channels and send these too another Mac. There's plenty of software around to do that. If you can start a stream on REAPER/Windows, your Linux box can receive it. That should be ample bandwidth for 64 channels 48/24.Īnd audio streaming is cross platform. OTOH, if there is nothing else on that network (PC-to-PC), without even a router or a switch in between, you can easily get 750 Mbps out of CAT6 with a less than 2 msec added latency. So you need a switch with audio-over-ethernet compatible QOS settings. And that's called QOS (Quality Of Service). It'll probably work at home and fail in other spots. And it still adds a lot to latency and chances of dropouts. Wifi 1.300 megabit, gives 300 Mbps one way, tops. ĬAT6 gigabit in practice is around 650 Mbps one way. Their Dante connector can handle up to 500 channels of uncompressed audio (per direction!) over one cat6 cable. Those have a special LAN socket instead of "normal" audio connectors like ADAT, XLR or SPDIF. Wifi transmission works very well for data where latency is not an issue but not for uncompressed audio that needs to be in absolute sync at 48k or even 96k. The restriction is, that audio-over-ip doesn't (currently) work over wifi due to timing problems (latency) and packet loss (caused by unreliable transmission/reception). For audio-over-ip you'll need some sort of interface, converter or maybe a plugin like ReaStream that allows to stream audio over your local network. For MIDI, this is very easy and at no cost by using rtpMIDI. The second part is using networking infrastructure to transmit audio or MIDI. But the connection is only graphics, no audio! So it's just for remote-manipulation. you literally sit in front of the desktop of the remote pc and can do anything. Ultravnc is a tool to remotely access the desktop of a pc (via lan, internet or wifi).
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