Hi ramses, many thanks for taking the time to reply to me. Much appreciated. It's only now that I've able to sit down and have a long look at this.
I do understand the concept behind routing. I've used many hardware mixers over the years and understand how to create submixes for the aux outs etc. But it's just the software side that I'm not 100% with, the relationship between the applications and the Software Playback channels in TM FX.
Thanks for the additional list - I shall have a look at these, although I have looked at many youtube videos on TM FX before posting, and I do mostly get it. But adding something like video conferencing into the mix takes it beyond the scope of the guides that are online that I've found.
I'll try to respond to some of your points...
ramses wrote:2. WAN (wide area network) between communication partner
In my case I don't believe that this is having a noticeable effect. My upload is about 10Mbps and the other guy's download is 15Mbps, and when we last tried this neither of us had anything else going in our houses using the bandwidth, apart from our phones (in stadby mode) syncing regular data, which would be negligible.
ramses wrote:3. a conferencing software where you do not know exactly how it works
I think I may have discovered something that might have a significant effect on the audio quality at the other end. I found this very detailed guide that looks like has only been in existence for a few of days... https://drive.google.com/file/d/1qGqDh5 … lY1_r/view Page 24 of this has a settings in Zoom called 'Enable Original Sound' (that need to be switched on at the zoom.us website before they are made available in the Zoom client) that looks essential if you're passing music through Zoom. I'll give that a go.
ramses wrote:4. You do not know or need to find out whether the communication software can use ASIO, if yes, use that.
From what I've found whilst searching, Zoom does not use ASIO. Also, the guide above does not once mention ASIO, therefore I'm assuming that it doesn't. Regarding WDM, I read the post you wrote on the other thread...
ramses wrote:<b>Non-ASIO aware OS / Applications -> WDM devices</b>
Windows and Applications who do not support ASIO need the WDM devices.
You create those in the RME driver settings, best only for those i/o ports which you require (speakers, phones, mic).
There is a TAB speaker, selec the I/O port which you want to use as default windows sound device.
Such an entry is created in the Windows audio settings.
There you need to make is the default sound device. Thats it for sound output under Windows and non-ASIO aware apps.
Unfortunately, after reading I'm unaware as to what I need to do. The first part of it seems to point to the 'Fireface USB Settings' utility, but then it seems to be referring to Windows settings.
ramses wrote:If Video and Audio does not work together, well then get rid of Video .. Video you could also perform over another medium like over Smartphone. Then you do not consume too much bandwidth.
The video cannot be moved from Zoom onto something like whatsapp because I need to send the Cubase screen over the video.
ramses wrote:You might also have to look whether you have a performance problem on your PC running both, DAW and the Video Conferencing software. Maybe you also need a higher ASIO buffersize. But then you get more latency in ... This is the way it goes and over WAN you have an additional Latency / Round Trip Time of between 13 - 40ms alone.
My PC is coping OK with resources. There's not much going on anyway - Cubase with mainly audio channels and very little in terms of plugins or VSTi. And Zoom doesn't consume much.
So perhaps the Zoom original sound thing above will help the audio quality, but I need to get the mixes right for me (between the Cubase main out and other guy's voice) and for him (between the Cubase main out and my voice). When I'm referring to "voice" here, I don't mean singing/recording voice, I just mean us two chatting over the music about what to do next. If I decrease my own voice in my own mix, then it decreases it for him as well, so I've something wrong somewhere. Here's what I have set up...
Windows - Sound Settings:
Main out - Speakers (RME Fireface UCX)
Cubase:
Main Out - AN 1/2
TotalMix FX Hardware Inputs:
AN 2 - I plug my mic in here
TotalMix FX Software Playback:
AN 1/2 labelled as 'Cubase'
AN 3/4 labelled as 'Zoom'
TotalMix FX Hardware Outputs:
AN 3/4 labelled as 'Dave' (the other guy) - Loopback enabled
AN 7/8 assigned to Phones 1 (PH 7/8) - Loopback enabled - I plug my headphones in here
To do my own submix I click on Phones 1 (PH 7/8) and give it signal from channels 'Mic 2', 'Cubase', and 'Zoom' - only those
To do Dave's submix I click channel 'Dave' (AN 3/4) and give it signal from channels 'Mic 2' and 'Cubase' - only those
Zoom - Audio Settings:
'Speaker' is set to 'Analog (7+8)(RME Fireface UCX)'
'Microphone' is set to 'Analog (3+4)(RME Fireface UCX)'
Is there anything in the above setup that jumps out to you as being wrong, given what I want to achieve?
Thanks so much.