progress report + some more questions

I´ve done some progress lately and thought I´d share some of it here, plus the questions I still have

First of all. Working with the “remote delay seconds” and upstream and downstream audio and video settings is crucial.
AND the way to do this, is to press play so that the bar graphs starts to show up. YES the manual explains this, but very vague, with the following sentence: “The two bar graphs show how well the two partners are synchronized during
playback and recording”

SO, that means, press play so that the bars show up, then experiment with different settings and make sure that the bars are high up. Basically you can increase the “remote delay settings” to compensate for bad connection, and/or higher video and audio settings. Also, sometimes when you change settings, you need to press stop and then play again to find out how the change really affected the system, because sometimes after a change of settings you will get inaccurate, or at least temporary, bar displays. So press stop and then press play again.

Second crucial thing, how to monitor. The manual says, to not use the “channel monitor” function, as this cause a delay, this is correct. However, what it fails to explain, according to me is how to then hear the performer.

This is how: Go to Devices --> vst connections - choose the studio folder - and on the “monitor 1” tab - choose your soundcards output. Now you can hear the perfomer

To be honest I still haven’t had the chance to try to record anything since I found these things out, so I’m still curious about if things are going to work properly now. But I am very thankful for the progress I´ve made, and yes Steinberg can send me a reward for figuring these things out :slight_smile: Here are some questions I still have:


what are the recommended setting for these things ( device setup --> vst audio system), and how do they affect things?

Activate multi processing

Activate ASIO guard

Activate Steinberg Audio power Scheme

Adjust for record latency

How will the “buffer size” on the performer’s side affect things? I understand it will be harder to play with a higher setting due to latency, but will this latency affect the recording, if let’s say the monitoring is done with direct monitoring? will a higher setting then create higher stability? Will this setting in any way affect the sync of the recording?

Also, why, and when is the option “add talkback channel” greyed out? (as it is on my current system)

Which is immediately followed by “You can see how much audio data is available per half second on either end.”
I repeat once more how it actually works:

  • Studio hits start. Appearently, nothing happens on the Studio side for as long as the Remote Latency setting allows (one second default).
  • in fact, during this time, Cubase/Nuendo has started to “play back” internally, providing the cuemix stream to the Performer.
  • The Performer performs to this cuemix and sends his audio back to the Studio. This all has to happen without glitches within the time given by Remote Latency
  • now when the latency period has passed, Studio is ready to play back along with the Performer audio perfectly in sync
    As such, you are right to say that this is a crucial setting. But the bargraphs are there to help; each shows 1/2 second of time and the bar indicates how much is available (at 50%, 0.25 sec audio are available).

Exactly. The Remote latency setting is the first one to look at when there are dropouts.

That is correct and you are right that this should be explained better in the manual. But it should “just” work when you just enable the Control Room (which I think Create/Repair VST Connect does automatically).

While these questions are beyond the scope of VST Connect and the items are explained in the Cubase/Nuendo manals, these are somewhat special and if you don’t know what they do you are advised to leave them in their default state, especially if all works well. All of these are for improving audio performance in certain cases.
ASIO Guard used to cause problems with VST Connect but afaik it does no longer with the newer VSt Connect versions. You may try to disable ASIO Guard if you suspect it to interfere with VST Connect and it should do no harm other than possibly reducing playback performance. The others I would leave on their own.

Basically, it should make no difference. However too low a buffer size may cause the system to be stressed (both Studio and Performer) and due to the additional load for audio and video transmission it might work better with higher buffer settings. If the Performer uses direct monitoring it is probably a good idea to raise the buffer size (1024 or more). OTOH, the Performer application is pretty straightforward so there shouldn’t be much stress so unless the Performer has a very old system, buffer size shouldn’t matter.

Because you already have one. There can be only one Talkback channel.

Thanks for the response musi. I actually had my first successfull session now. we did not have that much time, but it seemed to work and be in sync. I’m very happy! Despite that I would still like to ask some follow-up questions:

you wrote this:
You may try to disable ASIO Guard if you suspect it to interfere with VST Connect and it should do no harm other than possibly reducing playback performance. The others I would leave on their own.

What do you mean with “reducing playback performance”, reducing, what, in what way?

also, you wrote “the others I would leave on their own”, what is the default settings for these? at the moment I have:

Activate multi processing - on

Activate ASIO guard - off

Activate Steinberg Audio power Scheme -. 0ff

Adjust for record latency- on

Are these settings considered to be default? would you recommend having asio guard on, if it seems to work when I have it on?

some more questions. When I route my mic to the talkback I know that I can press the “talkback” function and then my perfomer hears me, however how can I tweak things regarding this area.

For example, I don’t hear myself talking in the mic, which makes things kinda weird, since the talkback function lowers the volume of him, plus I don’t hear myself, so it creates a very lonely and silent experience whenever I want to talk. I created a channel and routed the input to my talkback, when I turn on the monitor for this channel I will hear my talkback mic, but when I press the talkback function, this turns this off regardless

I would imagine that I would like to hear myself in my headphones when I talk to my perfomer, plus I would also like a less reduction of my perfomers microfone, because I would like to be able to have a normal conversation with him, which I felt was a bit hard, because, whenever I wanted to say something I had to hit the talkback button, but then quickly turn it off, so I could hear his respons, or his response would be very silent. Is there a way to set this up so the levels between him and me are more even?

Also, I haven’t really understood how the rehearsal button is suppose to work, would you mind explaining how that is supposed to work?

In the future I can also imagine having my perfomer having 2 channels hooked up, a microphone on one channel, for communication with me, and his electric guitar on the other. Then, whenever I would like to talk to my friend, I would have no trouble if the signal of the guitar went down, in fact it could be good, whereas I ofcourse want his talkback mic (channel 2) to be fully on, so we can talk. is there a smooth way to set this up? perhaps when I dont have talkback pressed his microphone would be slighly lower, so I can hear is guitar signal more clearly without the microphone picking up room sound from the string of his guitar, or perhaps even other noise from people in the house. at the same time, his mic should never be completely off, or it will be hard for him to communicate spontanously with me

Also, please include a guitar amp simulator in connect perfomer - next release ( both for pc, mac AND the ipad version) and please don’t force us to have to buy another guitar simulator plug in, or use a freeware version. It is so much hassle having to download it, and then putting things in the right folder and bla bla. Especially since I want to be able to make things simple for my none-computer-geek performers, ( cause as it is now it basically means I will have to visit them and download and intall everyhting for them - its too hard for my performers) thus having the guitar amplifier built in would be soooo much better. ( with the ability to record a clean signal, which you said will be in the next release). It does not have to be fancy at all, just so that we can actually record normal rock music, with is not possible without distortion I would argue

It is a bit beyond the scope of this thread, you should read the manual.
I guess default is ASIO Guard on, Multiprocessing on, adjust rec latency on and power scheme off. I always engage the power scheme on Windows machines, it may improve audio performance (more channels, fx etc) and the same is true for ASIO Guard, so if it works and causes no problems, enable it.

Yes, there are ways to do this and set the dim volume and also set how the auto-off Talkback function can be adjusted. But again this is all Control Room (Studio) functionality which is explained thoroughly in the manual and for everybody recording somebody else, whether next door or far remote should familiarize himself with the control room, it is a mighty tool tailored for this purpose and that is why VST Connect has been integrated there.
As for hearing your own voice: you are right to set up a channel of your own, there is no provision for that because usually the Control Room engineer listens via speakers and that would cause potential feedback.
Finally, you can assign a key to talkback on/off and make it latch etc. Read the Control Room manual section, it’s worth it even if you (like me) avoid manuals if possible :slight_smile:
One more thing - you can’t set the listening volume for the Performer as you said. This will be added in the next version.

It overrides the automatic switching of the talkback function. When it is off and the preference is set so, as soon as you start playback talkback becomes de-activated. This is because usually the Performer doesn’t want to be confused by sounds and talking from the Studio (Control Room). In situations like coaching however it may be desired to leave TB on all the time, this is what the rehearse button does. Keep in mind however that there is a long delay (as set by the Remote Latency parameter) which can be quite confusing. So for instance when you sing along to what you hear it will be significantly out of sync for the Performer. So usually you will want to leave everything as it is (Rehearsal off).

You can control which channels are recorded and which are not in the “Record” mixer section in the VST Connect plugin.
With the PRO version you can also control which single channels are recorded on the Performer end for HD transfer.

Sorry, this is again beyond the scope of VST Connect Performer which is a free tool. We have provided Inserts so this is possible, even compressor and eq are provided by default, so for guitar processing and the like you will have to live with the Inserts support.

thanks again for all the responses

About adding guitar processing. Sorry I just can’t drop this request just yet and thanks for listening to my arguments and questions. I just think that just as steinberg added a free reverb, since its very convenient to have when you record vocals, what is the difference between that and having guitar processing, I mean for me its absolutely crucial, The guitarist I want to record is a father of small children, and a wife, he can’t use a regular guitar amplifier and put a mic to that, its has to be done in the computer software. So why provide reverb for vocals, when not guitar processing for guitars. its not that we users are going to want you to add a ton of different inserts, its just this one thing. Is there even inserts for ipads?

Another problem has arisen. We started to get disconnected. what is the reason for that, any way to prevent it?
There should be no problem with the connection, we use skype all the time, and never get disconnected.
does for example low buffer size increase stress on the ipad and then increase the risk of causing disconnections?

want to raise this once again

Another problem has arisen. We started to get disconnected. it’s no crash, my performer just disappears, and get’s disconnected. what is the reason for that, any way to prevent it?
There should be no problem with the connection, we use skype all the time, and never get disconnected.
does for example low buffer size increase stress on the ipad and then increase the risk of causing disconnections?
What other factors can cause disconnection? We use enough latency seconds to have no load on the bars.And everything is in sync, so I dont understand why it suddenly disconnects.

What exactly happens, what are the messages that you see?
The actual connection has nothing to do with buffer sizes, Remote Latency or the like.

I dont remember exactly, I think its something like “connection lost” or something
He just suddenly disappears, its pretty discrete actually, suddenly he is just gone.
I just then called him up, he punched in a new code, and we were connected again.

If you need word for word what it says, we need to try it again first, so I can write it down

he uses an ipad, which he has bought a device which enables him to connect to internet through a cable.
but we also tried over wifi with the ipad, If I remember correctly, it worked to record with both connection types, so that was a good thing, however both connection types also got disconnected after a while. very weird