Hi,
we plan to do live streaming discussions using vmix call with 5-6 remote participants and two people in the studio. How can I have a sound engineer control the audio settings (levels and EQs) during the show on a separate controller or PC?

Our concept is that one person does the video mixing part, and another person deals with the audio. We need to do EQs on all callers, their microphones and rooms are just too different. And I do not want the video operator having to deal with that during a show as people call in right before the show. I did not find a way to have a remote Midi controller control the audio input EQs. My idea was to have some midi buttons to select the channel, and then use midi encoders for the EQ-faders (using some AKAI or Behringer MIDI controller). But we'd like to have some visual feedback (=EQ window of the selected audio input), too. I was looking into finding VST EQ plugins that can be controlled via MIDI, but I don't know how I could then select the right audio input channel?

How do you do this? Trying to route all audio to an external audio console or software on another PC, and then back to vmix, will make things very complicated....

Thanks for any suggestions!

Morki

doesn't look like vMix shortcuts extend to plugin controls, would be a good feature to see added though..

besides the external mixer option you could maybe try virtual soundcheck ?

to do that you ask each person to call you ahead of time and record their input.. then load up those 5 or 6 recordings and apply EQ and set levels.. once you have the mix adjusted using the recordings you replace the video files with video call inputs ready for the event.. the only problem with that is if anything changes during the call you may still have to make some changes on the fly.. also you would need the recordings to be made with each caller using the same equipment in the same location. its not perfect but if you can only use one person producing everything it does enable some preparation..

there are other options for separate audio producer but it gets a bit complex.. you could route all the callers to vMix busses, then use virtual audio cable software to route the busses out of vMix to a separate audio program for mixing (one that can control VSTs via midi).. you then route audio back into vMix again via the virtual audio cable as a stereo source.. you would need to use an extra computer monitor to show the audio mix on one display and the regular vmix interface on the other so each person has their own screen.. bit complex but in theory it should work as long as you setup different input controllers for each operator..

if you wanted your audio producer on a totally separate PC it is the same concept but you route the audio via ethernet to the second computer and then back to vMix using something like Dante Via Virtual Soundcard..

You could try the other way around... Use a MIDI device / Stream Deck on the same, or Panel Builder on a 2nd computer for the switching.

Hi mtone,

thanks for the explaination. I always thought vmix does noch have enough buses, but I did not think of duplicating the number by using them as mono channels.
I think your concept should work.

pros
- auto mix minus works
- freedom on all audio mixing by using external software/audio console (i.e. using dante gives full flexibility). I will need to look afor an easy to use solution. Any suggestions? I need 8-10 inputs (i.e. via Dante virtual sound card drivers), at least 7-band EQs for every channel (better: 12+ bands) or the ability to user VST plugins on every channel, and ideally even midi support... like I wrote before, assigning channels to midi buttons for channel strip selection, then a bunch of encoders/faders for the EQ.

cons
- complexity
- cannot send the improved audio to the other vmix call participants


I'll look for a audio mixer software and to some tests next week. We try to keep all our stuff very compact, so I'd prefer just a laptop and some midi controller instead of a real audio console...

what annoys me a little: vmix can do everything I need (EQs+VST plugins on every channel), but I just cannot access them during the show. If there only was a possibility to have midi access to the individual EQs and ideally a detachable audio window showing those EQs that I can move to another screen, we'd be ready to go....

If you un-dock the sound mixer, then it can operate on a second monitor/display.

So whether you're connecting remotely, or using it in person, just connect to that separate display to adjust all the audio settings, without messing with the main vMix operator.

So for Separate Aufdio and Video Input you need to set direct out audio signals out of vMix.

These Audiosignals can be mixed by the audio Operator and then send it back to Vmix.

I will investigate how to do the direct out. The reInput is done by adding addition Inputs with audio-only inputs from soundcard. Backside you loose audio follows vmix in the first place. Iam not sure if you can solve this by creating virtual inputs of your calls and reassaign the correct processed audio input.


Second solution is a bit more uncommon. Use Waves audio plugins, and here the Studio-Rack Plugin
(it is ment to send audio Streams to a waves server process them there re insert them in your audio track witch is exactly what you want) but at least you will need one waves server. But you can route audio also directly from computer to computer (in v9 StudioRack routing was able without waves server hardware) in the current v11 version a waves server is required as a hardware dongle.
The audio engineer can also use an mixing desk with waves card for audio routing.

The Waves stuff it self is live proof when set up right.
The StudioVersion (for DAW use) needs propper set up and testing on the computer, in fact it is an complex driver system.

Have fun testing

OK Thank You, i allways forget that the input mv stacking also works with audio inputs and is the way vMix is supposed to use.

to OP:

Workflow Solution:

All mics on location go directly to the audio desk of the audio engineer, he processes them an make it availible fpr either as post fader direkt out via separate audio siganls you can either grab by an audiointerface with huge ammount of line inputs (vMix supports multiple soundcards to host but ona soundcar is better. also consider network audio drivers like DanteDVS or WavesSondgrid) Alternetivly the audio engineer cold do a location submix for you.
WIch works better is dependent of how manny diferent sub studios you have (e.g. weather man in news studio) and hao often you switch them and how fast. For a talkshow like enviroment a submix should be convenient.

You said you want use 5-6 callers wich is the maximum ammount vMix can handle out of the box for external sound processing (real max is 7 but one bus for internal videoclip prozessing should be left). Use Audio busses A to G to assign ech caller to a separate bus. Each bus is set to an individual audio output, this are the inputs for your audio engineer wich he can process. I think calls should be audio auto follow, so the sound guy shuld send you back each caller as an seperate input.

Mix processed call audio and call video with the input mv like mtone said.
Mind the roundtrip latency so audio will remain lipsync!

Now you should be able to split audio and video on separate oparators. Figure out ho to work wether with submixes or with autofollow separate audion ins is dependent on show setting and the timing. when it gets to complex and fast shows you will spend the same time programming in vmix than building intercom and define cues and rehearsing them.

have fun

Hi Paul,
thanks for extensive descriptions. I'll try to go that way and probably use Dante (definitely not going a D/A-A/D route). I'd prefer to do the audio mix on a Laptop instead of a console since we want to keep our setup as flexible and portable as possible, so we'll do the first shot using soem mixer software. Any suggestions on another PC based audio mixer/console software?
Regarding the limit to 6 Buses: shouldn't I be able to extend this to 12 by using them as mono channels like mtone suggested in #6?

@Morki:

Using Dante DVS on Both Computers to enable DVS to DVS Patching wont work direktly. This requires a Third „Dante Hardware“ Device wich will work as the clockmaster. Dante protocol requires a clockmaster ti work and dante disables dvs to provide clock because of unreliability. But with a third dante device it will work. And you are able to route from pc to pc directly.

There is Dante Solution for that called DanteVIA. This was designed to grab audio from any application wise. So you can assign chromebrowser to output channels 1&2 and zoom to 13 as an example. Keep in Mind that tha Channel count is limited to 48 total. But then Internaly splitted in 16Channels of ASIO (your Mixer Software will only access 16channels i/o) and 32 Channels for applications. But i never worked with DanteVIA in practical.

Also keep in mind that you end up in huge delay with dante dvs. Dante DVS has a network delay of minimum 4ms + ASIO Buffer per direction. For processed caller audio you will stack this ammounth four times.

What i did is pc to pc routing with waves soundgrid/StudioGrid. Which is also a bit quicker with 2,1ms Roundtrip per computer + 2xASIO Buffer.
But with studiogrid v9 current version is v11 wich i never tested. In v11 some features are locked till you run a waves hardware server in the network for sales reason.

Also my personal opinion on kepping flexible. A 19inch mixer is not bigger to transport than a laptop with a decent motor faderboard and audiointerface. An it is more reliable.

The Mixer software you use is new to me. I used in theatre applications ableton live a lot, and it is great for combine jingle playback and live mixing with external inputs.
I also know that waves have the LV-Emotions mixer as a software wich is quite expensive but focused on replacing a mixing desk, but works best with a touchscreen.

Other Software thar are tuned to live processing i dont know so far. Most DAWs will do the job gor shure but are not developed on live mixing reliability.

@RDP:

How do you manage to create 12 calls simultaneously. The biggest vmix license is limited to 8callers?
And you did archive the mono split by the routing matrix in vmix right?

Greetings paul

@Paul:

* I'll be setting up the system next week and did not realize we are limited to 8 callers total (which we'll have to live with).
* Also, I haven't tested splitting the buses into mono but hope that works as intended so was asking if Moriki had successfully gotten it to work yet.
* I'm currently planning to interconnect vMix with the studio mixer and IO with Dante DVS by splitting buses as discussed before (will also travel through a Waves Server for additional processing). But, if there is a better way to get the caller's isolated audio out of vMix I would consider that as well. I'm currently reading through the NDI capabilities but don't see a way to pick up the direct outs from each call. Do you know if that's possible?