[DaPoParticipant]

Sad to say I’d actually seen that, thought it was good, and even laminated it for other users! Kind of *recognised* the function as something I’d seen. I didn’t recall it when needed, and I was too lacking in cool to reschool myself. Funny that they call it an “encoder” there – elsewhere it seems to be just the touchscreen rotary.

[DaPoParticipant]

Coming back to report success after another visit to examine this problem.

Initially tonight I again had the problem described above, then I realised that the power amp was turned up to maximum.
We have two groups of people who use this site, and the other group has only one pathway – to “L-R” – that they need concern themselves with. They habitualy turn the power amp to maximum, then reverse-adjust the mixer so that its pre-amps have little work to do. Not much signal, then, going directly from the desk to the USB output.

My error must have had to do with my initial lack of success with getting an audible mix through the matrix onto the USB-based recordings I was making. That, in turn, was due to my inability at the time to adjust the components of that mix (see my other post). I think I then reverted to making do with the USB receiving simply from L-R. That placed constraints on the ambient pickup microphone.

An abiding difficulty was doing this alone, with long walks between tests of inputs, and the slow approach of making recordings on the PC receiving the USB output in order to assess quality. I realise that at one point I was being misled about poor level quality just because I was playing back output from there through the house speakers, L-R, and not realising that the gain had been set low on that SQ input.

I think the key lesson for me here is to use the monitoring tools that the unit comes with. The headphones to hand needed an adaptor to work with the socket on the SQ-5. But I would have done well to have good, well-insulated headphones and a strong understanding of how to test the sound at various points. “PAFL” – it should be “baffle”! The settings screen currently on page 78 of the manual (12.7 PAFL) certainly requires a visit to that page. And what a pity we resorted to Latin at all for our listening standpoints! ‘Pre-‘ and ‘Post-‘ both begin with “P”, so why not go the whole way and call them “BAFL”! Before and After Fade … anyway, I digress. I acknowledge this terminology gotcha is bigger than A&H’s purview.

[DaPoParticipant]

I realised my problem. I had been reaching for the Gain knob (top left) instead of the ‘touchscreen rotary’ which is to the bottom right of the touchscreen. The window turns to yellow when touched, and the relevant knob then also glows yellow, so I suppose that’s a clue. (There is, of course, a bloom of coloured lights when you push the level window.)

[DaPoParticipant]

I was going to reply here for two reasons
You’ve covered one of them now! You were summarising para 4, and para 4 made sense, so I figured you meant “post fade”.

Some advisors suggest that a matrix works ok by and large as a compromise. But your point about the loss of individual channel control is true.

What interests me from your explanation of aux control is that post-fader from the source inputs work by relative displacement, not in an absolute way. In other words, if I’ve set all channels in my aux mix to unity, but there’s a spiky profile on L-R, I will get a different result from if I’d started with them all at near the bottom. Is that so? There will be no initial “copy” of fader levels; rather, the levels will alter by some metric increment as you play with the faders at L-R. Now, let’s say in pre-check, you’ve boosted a weak channel on your aux. We “mute” on L-R by sending the fader to minimum. Will that mean that a faint signal will still come through to USB? Notionally, the fader will have dropped by the same amount, but not to nothing.

I’m interested because I have a big acoustic instrument (a pipe organ) that I don’t want through L-R but need at strength in the broadcast mix. I’ve used a matrix in the past for this, assigning the “ambient” channel a special group for the job. To use this via aux bus, I’d simply have an additional contributing input over the ones assigned to L-R.

Tks

[DaPoParticipant]

Hi, volounteer – sorry I was so slow getting to see your reply.

We may yet also livestream on a conventional service, but we use Zoom for two-way interaction. Some participants are remotely located. We see them on a projected screen, and sometimes hear from them through the church’s PA, controlled of course via the ZED-428.

I liked the idea of the matrix rather than a complete reworking via an AUX, because we’ve got just one operator.
If the mix of voices is right for the ‘house’, we’re happy for those relative levels to serve for Zoom as well.
But for the external delivery, we need to capture the ambient sounds that usually go ‘un-micd’ in the building – principally the pipe organ – which vocal mics won’t do well.
If we use the matrix, we can mix in the organ to the already-established voice mix.
In other circumstances – with two experienced audio techs – perhaps in different rooms – I agree that porting all the sounds might be the way to go.

[DaPoParticipant]

Mike_C, thanks so much for that really thorough and complete-sounding advice – and sorry that I’m so late reading this.
I was relying on the “Notify me of follow-up replies by email” feature of this forum. (If it did that, I didn’t see it.)

I see then from your advice that it’s OK to use the “M” mono output mix for our Zoom-only livestream.
That rids us of worrying about the odds-and-evens thing with the numbered groups.
Thanks, though, for covering that contingency, too, which we can remember if groups get used in the way you describe.

My remaining question with the Matrix (on the ZED-428 anyway) is the other pairing that we’re dealing with: the USB output is preconfigured to accept both Matrix “vectors” I guess – both 1 and 2: “MTX1-2”. At present, I’m advising that we pump signal through both column 1 and column 2. Whatever we do with the knobs in the first column of the matrix, we copy to the second. I’m assuming that the signals are simply summed together. If, instead, the available signal output is allocated by proportion, we could just ignore, say, column 2 altogether (maybe set its master level to zero) and concentrate on level-setting in the first column. Do you have any opinion to offer here?

Yes, we’re receiving incoming audio from the viewers – that’s one reason why we’re Zooming rather than simply one-way livestreaming. On this point, we seem to have a duality. We can receive the USB input as “Playback to LR“, or we can have it come in as ST3 (if I recall the identifier correctly). We seem to have got more sound by using the former. There is more control (via fader, etc.) using the latter, coming through the channel strip. We’ve often got both of them active – ‘belt and braces’ – but that’s not good when you want to control things – e.g., to mute. Two places to look. Is there a best practice here?

I promise to visit this page again this weekend and/or soon after, and not rely on email notifications! 🙂

[DaPoParticipant]

(Sorry: that was my first post here. Assuming I can’t edit my post, some corrections on my terms:)
I’ve used “channels” a few times where I meant “groups”.
The “paired groups thing”.
Panning as I understand it will deliver signal in the assigned relative proportion to Group “1” or “2”, for example, in the same way as it would to “left””and “right”. I think my counter-intuitive discovery was that Group 1 signal improved when I panned the contributing channel Right (“Even”) rather than Left (“Odd”). Which is odd!

[Author]

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