Not owning an Apogee Ensemble, maybe someone else who does has more definitive info can jump in, but do not confuse the lower level of the SPDIF input with "lesser quality". It's a digital connection, with no analog conversion.
-18 dBFS on a digital scale is the same as 0dB on an analog scale.
I think what you're experiencing, is the fact that so many people think they need to get the meters in Logic closer to 0dB, which is actually NOT what you want (when staying ITB). This thinking is left over from the days of analogue recording, or the early days of 16 bit digital recording.
Turn your other sources in Logic down, and you'll reap the benefits of not overloading the 2 buss, allowing the plug-ins to do their computations without distorting, etc... Then you can bring the over all level of your mix back up, in mastering.
If the Motif is considerably lower in volume than that, you could always insert a gainer plug-in on the audio track.
As to the 15 minutes, that is what Logic has pre-allocated for recording to your hard drive. It "re-sets" each time you go into record, so it's not saying "you only have 15 minutes of record time available". It's saying, "you have 15 minutes of hard disk space allocated each time you go into record, based on your sample rate and bit depth settings". This can be changed in the Audio pathway, but it's recommended to leave this number as low as possible/necessary, to avoid disk fragmentation.
Ok, i'll have to check on what the db reading is...where do i do that. is that the number above the level indicator on the bottom left side of logic that changes everytime i hit a note? i think i saw a lot of -27 or -35 or -42. so that's still not -18.
i'm just so use to being able to adjust the levels...not sure why i can not adjust the levels at all on my input? is that the way SPDIF works??? it just is what it is going in and you can't do anything about it?
let me know if there is anything i should be able to adjust to get the initial input up higher or lower if i wanted to. Thanks
The thing to remember though, is the SPDIF is a digital input. there is no analog signal from which to amplify. Thus, you generally don't see a level control on a digital input.
There may be something in software, so consult your Apogee manual (or wait till an Apogee owner jumps in), but I personally haven't seen anything like that on any of the digital inputs of the gear I've owned. The volume was controlled at the source.
yes i configured everything in Maestro. I looked last night and didn't see anything to adjust volume for SPDIF. I did look at the route mixer, i did see faders for the Input 9/10 (my SPDIF inputs) but me adjusting them did nothing at all. let me know what you have your settings in maestro for.
here's what i have.
yamaha motif xs8 (connected for ensemble via spdif)(will hook up midi when i get midi cable)
avalon 737 preamp (plan to plug into the trs input for "return" on analog in 1 or 2 on ensemble, unless you suggest something else.)
i own two condenser mics that will be put into the avalon preamp
Let me know and i'll readjust everything b/c i actually plan on erasing everything and reinstalling everything once my usb cable comes in.
the mac screen froze up big time once i installed apogee driver plugged in firewire and then turned the apogee ensemble on. the screen went haywire and i had to do a "hard restart). maybe you know what happened and why it did? Thanks
yes i configured everything in Maestro. I looked last
night and didn't see anything to adjust volume for SPDIF.
There is no (never should be) any adjustment for in incoming
s/pdif signal. First of all, it would compromise the data.
The whole point of a digital transfer is a bit by bit copy, adding a software volume control will ruin the integrity.
"It's a digital transfer" not analog.
If need be, raise the volume after the transfer.
The better method in Logic and most other DAW's is to not
modify the waveform by normalizing or raising the gain
with the sample editor, but simply upping the mixer level
for that track.
To amend my previous post.
There is no (never should be) any adjustment for in
incoming s/pdif signal.
There should be no level adjustment period, if there is it's not a direct digital transfer.
The whole point of a digital transfer is a bit by bit
copy, adding a software volume control.
Or hardware volume control, s/pdif transfer should not have "adjustments".
Hey, let me know when you get home how you have all your stuff setup as I see you also have the same avalon preamp i have but have not setup yet. So, we have a lot in common. I do Gospel R&B. My wife and I are both vocalists as well so we're hoping to be able to work the avalon pre with the neumann u87 mic into apogee ensemble and get that radio ready vocal sound. know what i mean.
so, on top of the apogee ensemble setup, please let me know how you have your avalon setup and what your settings are on there, and how you get the sound into logic through the apogee.
look forward to hearing from you.
Can you explain the whole +4 Pro vs -10 Consumer levels.
in Maestro, I had an option to switch b/w +4 and -10 for the analog inputs and moving it to -10 made my analog input to logic louder.
however, the spdif channels have no such option in the maestro mixer for apogee ensemble...and all my outputs max out at +6 in motif.
Please educate me on this. i would think +4 would be louder and better, but it appears to be the other way for analog
The difference is simply levels.
+4 pro level is louder by about 12dB than -10 consumer level. the labels "Pro" and "Consumer" level have NOTHING to do with the audio quality. Keep that in mind.
When you press the Apogee from +4 to -10dB, and you are feeding it the SAME signal from the Motif, of course, it will become 12dB louder. this is because you are feeding an exremely loud signal, and then making it louder with the removal of the singal padding in the apogee.
The reason the Apogee has this level difference, is that, when you are feeding it a +4 level, it pads it down to acceptable levels, to match the -10 level. a lot of gear does this, simply to not have to incorporate two circuits to handle the different levels in every channel input. The sound quality will not change.
You should read this, to understand the dB, and it's various ways of representing it: