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Dante PCIE + VEP + Cubase?

jamwerks said:
What buffer settings are you able to use on you daw & VEP?
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Depends on the complexity of the music, but it fluctuates between 256 to 1024. Obviously, the more instances I connect, the higher the buffer requirements.

I think most VEP instances are set to 1 or 2 buffers, can't remember off hand.
 
Mishabou said:
Raise instance being one, but there are other nice features not available when you use VEP in stand alone mode.

I have a few hardware synth routed directly in CB and I also use the JL Cooper Master Fader Pro for CC, Roland A88 controller, iPad Pro and an Avid S6.
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Curious if you've tried using external instruments to route audio and midi via dante to your VEP slaves?
 
benatural said:
To make sure I understand what you mean, are you saying you monitor your slaves that have VEP on standalone through Dante routed to Pro Tools on your master? Do you write in Pro Tools too?
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I don't have Dante but if I were starting over I might go that way. I used to have five PC satellite computers playing samples back but am down to three now, with two Macs. Mac 1 is my DAW with Digital Performer and Finale (and Sibelius and probably Dorico at some point).

Midi is on MidiOverLAN and audio is all hardware with RME and MOTU.

Mac 2 is more just a print-and-mix computer that has Pro Tools on it. I monitor through PT while composing and then once I'm ready to print and mix I raise the buffer from 64 to 512.
 
I run both Dante and VEP, but for completely different purposes. This thread appears to conflate them. You CAN use Dante as a way to transport audio between sample servers and your DAW. In this way it might replace MADI or ADAT or something. That is not what I do.

I use VEP to connect two sample servers to the main DAW. MIDI and the sample audio all stream across a dedicated Ethernet network. This is physically separate from my Dante network.

On a separate Ethernet switch, I have a Dante network. It routes about 40ch of audio around the studio. The main outs of my DAW, for example, go out via Focusrite PCIe card to a Focusrite D16R, which feeds AES directly to my Genelec surround system. All monitor feeds in the studio are distributed via Ethernet to Focusrite AM2 boxes. It is very convenient. I can route any channel of audio anywhere, or to multiple places at once - this is how I feed my Midas M32 console that handles headphone mixes, and misc. audio routing duties. The PCIe card has been rock solid for me, and Ethernet is a delightfully easy way to move 128ch of audio anywhere.

I have a separate Internet network that some machines are on, but not all.

Dante is the audio backbone of my studio, but everything related to samples runs through VEP in the standard way. I have about 600 tracks in the template at the moment, and it works quite smoothly.
 
Nathanael Iversen said:
I run both Dante and VEP, but for completely different purposes. This thread appears to conflate them. You CAN use Dante as a way to transport audio between sample servers and your DAW. In this way it might replace MADI or ADAT or something. That is not what I do.

I use VEP to connect two sample servers to the main DAW. MIDI and the sample audio all stream across a dedicated Ethernet network. This is physically separate from my Dante network.

On a separate Ethernet switch, I have a Dante network. It routes about 40ch of audio around the studio. The main outs of my DAW, for example, go out via Focusrite PCIe card to a Focusrite D16R, which feeds AES directly to my Genelec surround system. All monitor feeds in the studio are distributed via Ethernet to Focusrite AM2 boxes. It is very convenient. I can route any channel of audio anywhere, or to multiple places at once - this is how I feed my Midas M32 console that handles headphone mixes, and misc. audio routing duties. The PCIe card has been rock solid for me, and Ethernet is a delightfully easy way to move 128ch of audio anywhere.

I have a separate Internet network that some machines are on, but not all.

Dante is the audio backbone of my studio, but everything related to samples runs through VEP in the standard way. I have about 600 tracks in the template at the moment, and it works quite smoothly.
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Thank you for sharing your setup. I think I'm going to pull the trigger and give it a shot.

If you don't mind me asking, what switch do you use for your Dante network?
 
RE: Dante Via - it is not low-latency. I use it for playing Spotify and audio from the Web. It is not for latency sensitive applications. Dante is not multi-client on PC, but then most things aren't. Not an issue for me, as I'm either in Cubase or in Dorico, but never both. But could be a deal-breaker for some. Technology. Many horses. Many courses...

Oh, and for Ethernet switch, I use a Linksys LGS116P. This is supports POE (Power over Ethernet). This lets me run an Ethernet cable to the AM2 headphone monitor boxes, and they get power and signal from the same Ethernet cable. Very slick. Any GBE switch will do for Dante, however.
 
There must be some reason technical reason for Dante still not being multi-client on PC. FWIW, RME, Lynx and many others are multi-client. I often have Cubase, Dorico & PT open at the same time, and need that functionality. I went the RME route though I'm sure Dante is the future.
 
Nathanael Iversen said:
RE: Dante Via - it is not low-latency. I use it for playing Spotify and audio from the Web. It is not for latency sensitive applications. Dante is not multi-client on PC, but then most things aren't. Not an issue for me, as I'm either in Cubase or in Dorico, but never both. But could be a deal-breaker for some. Technology. Many horses. Many courses...

Oh, and for Ethernet switch, I use a Linksys LGS116P. This is supports POE (Power over Ethernet). This lets me run an Ethernet cable to the AM2 headphone monitor boxes, and they get power and signal from the same Ethernet cable. Very slick. Any GBE switch will do for Dante, however.
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Very good to know. Does VIA when used with Dante hardware introduce latency to all Dante channels, hardware and virtual? Or is it just apps and devices that use VIA, and non VIA Dante hardware etc remain unaffected? If the latter then that will work for me.
 
VIA only. The hardware and ASIO driver are what give low latency. VIA is just software and doesn't have dedicated hardware.

I don't have VIA on my DAW. I run it on my laptop where email, etc. live. VIA is how the laptop connects into the studio infrastructure. I don't use Youtube, Spotify, etc. on my DAW. DAW is production tasks only, so I never miss it there.
 
you may want to do a search to find the other threads where I have detailed my setup and motivations. I did Dante because I ran out of IO on an RME UFX. Once you get out of "interface + ADAT expansions", the options for low latency monitoring in particular, narrow significantly. I solved it for me in a way that uses Dante. But if I fit into normal interfaces, there is no question it is simpler (and cheaper). If you are running a multi-room facility, or have lots of audio moving around, the world looks different. That difference may or may not be important to what you are doing. I love being able to add more IO just by putting another box on the network. Is that important to your application? It may or may not be. Dante works. It is very reliable and fast for large channel counts. It is used extensively in live sound with massive channel counts. VSL uses it for orchestral recording. So it works, it is just a question of suitability for application.
 
My studio is using Dante, hardware wise with Focusrite. I love it, with some Cisco business switches I already was plumbed with by happenstance I set it up with traffic prioritization and get millisecond latency. Advantages are the latency and it just always works - no fiddling.

VEP would work but I don't want extra complexity, extra cost, new hardware keys and such. Dante at the moment is being just used for routing signal back and forth for recording, but I'm trying out using macOS NetworkMIDI + Dante as a replacement for VEP.

TLDR: Also, FWIW - I'm honestly skeptical of VEP's reported latencies. I was involved with the formation of the iee1588 standard at HP labs years ago, doing network measurements (data shuttling) is difficult because purely of latency due to the software stacks. Just knowing accurately what is your latency is difficult, it's more common to fool yourself rather than know what the latency really is. You need to scope it with a common 10MHz reference, just reporting on NTP and software stack ideas of time doesn't cut it. VEP are a bunch of software guys AFAIK, did they measure latency properly?

May not matter in practice, if it works it works well enough I guess, but I can't help but trust Dante over VEP with my background. I'll tell you, even in an ideal situation which is ASIC 1588 nodes and business switches with traffic shaping, you still can/do get glitches. With regard to clocks digital doesn't really exist in the real world (just an approximation of a digital clock).

FWIW
 
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In my view NONE of the products are reporting true end to end latency anywhere because of so many pieces from different players involved. The only way to know for sure the latency you are getting is to do round trip loopback tests with the software you plan to use.
 
Dewdman42 said:
In my view NONE of the products are reporting true end to end latency anywhere because of so many pieces from different players involved. The only way to know for sure the latency you are getting is to do round trip loopback tests with the software you plan to use.
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Dante conforms to 1588, its true latency.
 
From analog audio point to analog audio endpoint? Sorry I don’t trust any reported number unless you tested it yourself with RTL loopback
 
I'm an engineer by day and at the moment happen to be working on a distributed 1588 measurement system. This uses GPS and triangulation for <something> location (think microseconds), and for the comm backplane it's 1588. All I'm saying is, that if a product conforms to 1588 and you believe them, then the latencies are accurate. I've read the whitepapers that Audionate put out and believe they know their stuff. Otherwise I don't know what you mean by 'analog to analog', the 1588 protocol is network defined. If you have a ASIC solution like I do then the spec is to the FPGA (which has latency but in practice is ignored), of course thunderbolt adds latency outside of that if you're not hardware mixing.
 
That's what I just said :sneaky: ... anyhow we're off in the weeds. I'll report how well it goes doing a OS X network MIDI + Dante setup.
 
OK I've done a basic experiment using Dante as the audio transport and macOS network MIDI as the MIDI transport. Within the confines of the experiment (a couple channels) it works perfectly with no discernible latency. There's a number of pieces to put together, I'll sketch it out.

Setup
  • Setup Dante network.
    • I have 32 channels of hardware Dante on the slave side, but I need 44 for a standard orchestra (24 instruments in stereo). I could go mono but decided to just DVS (Dante Virtual Soundcard) on both ends with 64 channels total. My network is configured properly with QoS on the switches.
    • Just use Dante Controller to hook the outputs of the slave channels to the inputs of the controller/main computer
  • Setup the MIDI network channel
    • Go into Audio MIDI setup network panel and set up network MIDI channels. I need 24 so I created two sessions (16 MIDI channels per session) on both computers, then just had them connect to each other.
  • Setup the VI server.
    • There are articles on how to do this in Logic. One problem, Logic really likes to have a single audio interface, and a single document open at a time. With this I need AVS to be the audio device - not my hardware interface I normally use. Plus I didn't want this to interfere with the work I do on the 'slave' (it's really a slave to the other computer since it has the VI's, but I do most of my work on it).
    • Setup MainStage 3. Neat solution, MainStage supports setting the audio interface per document, unlike Logic, and is better with multiple documents open.
    • Create a new MainStage session, get rid of the junk in there, then create a bunch of "Keyboards" in the layout, but assign each one of them to one of the MIDI channels from the two network MIDI interfaces created earlier.
    • Then add channel strips for each instrument (piccolo, flute, ...), and set them up to have MIDI input from each of the 'keyboard' objects.
    • So basically piccolo takes your first network MIDI session channel 1 as it's input, flute takes the first session channel 2, etc.
    • Then just route their respective outputs to AVS channels (1-2, 3-4, etc).
  • Setup the master computer
    • This is in logic. Put down a bunch of "External MIDI" instruments. Route the MIDI to the network MIDI channels (piccolo goes to first interface channel 1, etc). Audio output goes to whatever you usually have.
Latency

I have three managed switches between the two computers which is the maximum recommended from Audinate
  • AVS
    • allows for a minimum latency of 4mS
    • In Dante that means it's the maximum latency, you can get better than that
  • Logic
    • 32 I/O buffer gives some glitches
    • 64 is glitch free in my basic testing with a roundtrip of 9mS
  • Dante
    • Dante Controller normally gives you actual latencies per connection/device, but it's not pulling it in for some reason. Something to investigate, I've gotten this before
  • Network MIDI
    • Audio MIDI Setup is reporting 0 latency but I don't know if that's real


Done! I only set up a few because to get the templates worked out will take some time as it's a lot of repetitive setting of stuff, but it can obviously be saved to a template. Best is I can leave MainStage up all the time, it won't interfere with the Logic work on that computer, and then on my other computer now I can get some work done and stream all my samples from the computer which has them.

Of course as said earlier this all could be done with VEP, but I'm already a Dante studio and like to stay with that if I can. I will go ahead and flesh out a full orchestra to see how it works with 44 channels of audio streaming with 24 channels of MIDI.
 
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