Level Meter Question: What am I looking at?

[darkecho]

http://i90.photobucket.com/albums/k266/darkecho_/Picture2.png

 

I have it set to RMS, and the bars look like they are measuring RMS..

 

but the values to the right of the bar (or top, depending on how you have your head tilted) dont seem to agree with the physical bar..

 

Are the values noting peak? (even though I have it set to show me RMS?

 

is there a better level meter for checking things in RMS? thanks!

[David Nahmani]

It's a hold of the max RMS value since you started playback. Just like what the peak detector is to the peak meter on the channel strip.

[darkecho]

the weird thing is that it seems to reflect what the meters in the mixer say, and those are all peak right?

 

the values at the end of the meters in the level meter plugin dont seem to change between RMS/Peak/RMS and Peak settings, they keep showing levels that are much higher than the RMS physical bar is showing, for instance, the picture I posted.. I cleared it a few times so that it was acurate..

 

the RMS blue bar never exceeded -10, probably not even -15, but the values are -3dBFS (which is closer to what the mixer window shows)

 

Is there a better plugin for RMS metering?

[David Nahmani]

Correction: it's a peak hold, you're right.

 

It's usually important to know what was the max peak value you reached, while it doesn't really matter what max RMS value you hit (since RMS is already an average).

[teta]

if you want a metering plugin that shows the numbers of your RMS levels also, you can get one for free.

 

check out Sonalksis FreeG. you can find it at sonalksis site:

 

http://www.sonalksis.com/index.php?section_id=99

[darkecho]

I am using an apogee duet and the techs said its converters are calibrated to -16dBFS

So I am using the RMS meter to get the levels of each individual track at -16dBFS Pre-fader.

[marcel72]

I am using an apogee duet and the techs said its converters are calibrated to -16dBFS

So I am using the RMS meter to get the levels of each individual track at -16dBFS Pre-fader.

 

What the tech meant is that Apogee hardware is calibrated so that 1.228Vrms=+4dBu=0VU=-16dBfs. This is a reference for setting the levels of I/O devices with respect to the standard that has been established for the operating levels of professional recording equipment. Digidesign hardware uses a -18dBfs reference, some other stuff uses -20dBfs...

 

This really has no bearing on the levels of tracks within Logic. This is just a standard for input and output levels, and would be used if you were setting gain levels on analogue gear that preceeded or followed your Duet in the signal path. For example, I set the head amps on my mixer so that the meters read 0VU with a -16dBfs signal coming out of my Rosettas and all faders at unity, so that all of the 24 channels are at the same level coming out of the DAW.

 

Hope this makes some sense to you. My point, really, is that you probably don't need to concern yourself with this.

[darkecho]

The duet outputs at -10, where does the +4 come from??

 

anyways... I was under the impression that I needed to have my plugins come into the DAW at a certain level which was governed by the input level (-16dBFS) of the duet..

 

would this be true when I am recording using a mic? to adjust the level to RMS -16dBFS which is optimal for the converters?

 

would it also be optimal to send a master output level at RMS -16dBFS to my monitors?

 

 

I think I was confusing giving the plugins enough headroom so I dont clip them internally... what level should I run into them with to avoid internal clipping?

[David Nahmani]

So I am using the RMS meter to get the levels of each individual track at -16dBFS Pre-fader.

 

That... doesn't make any sense? Why are you trying to get the levels of each individual track at -16dBFS? And what kind of signal are you using to calibrate? Do you realize your settings would be different wether you use a sine wave, rock music, classical music, white noise...?

 

The only thing you should worry about when recording is to avoid clipping. This is done by leaving plenty of headroom to your A/D converters by adjusting the Gain knob depending on the nature of the incoming signal: the amount of desired headroom will be completely different depending on the situation, the type of recording, the distance mic/source, the genre, the instrument, the musician playing that instrument... for example, if the source is predictable (ie a CD or DAT), the ideal amount of headroom would be 0 dB. If the source has a low dynamic (heavy distorted guitars), you could decide on 6 to 12 dB or headroom, whereas if you're recording an experimental singer or a jazz percussionist you may want more than 18 dB.

 

To adjust the amount of headroom, you want to look at the peak detector, not a RMS meter.

[darkecho]

I am mostly attempting to start off with an initial level that won't internally clip my plugins (prefader with the fader set to unity)

[David Nahmani]

If you're clipping the plug-ins that means you're clipping the A/D converter, and that's what you should be worrying about. Furthermore you can't clip the input of a 32 bit plug-in floating point (All of Logic's plug-ins and most 3rd party plug-ins are 32 bit floating point).

 

If you're using any 24 bit fixed 3rd party plug-ins, the only way to know for sure you're not clipping it is to watch the input level (with the plug-in's own level meter, or by inserting a level meter before the plug-in), and possibly turn down the gain before that plug-in.

 

But if you follow common digital recording practice (leave yourself 6dB of headroom or more) and don't go crazy with your gain staging (don't start inserting plug-ins that add 6 dB of gain or more) you should never have to worry about any of it.

[marcel72]

would this be true when I am recording using a mic? to adjust the level to RMS -16dBFS which is optimal for the converters?

 

would it also be optimal to send a master output level at RMS -16dBFS to my monitors?

 

No, and no. -16dBfs isn't really optimal for your converters, it's just a reference. 0VU is generally the optimal RMS operating level for a modern professional console's analogue mix bus, but you're not running one of those, so don't worry about it. When you turn your monitors up or down, that all goes out the window anyways.

 

AFA what you record, here's a better rule of thumb:

 

Try to get the elements of a multitrack recording to the level they will ultimately be at (relative to the other elements of this recording) as close to the source as possible, and then make as few gain adjustments as possible as you progress down the chain.

 

Don't record the returns from a reverb unit so they peak at -3dBfs, because you'll just turn them down about 35dB off the top. Conversely, don't record the snare drum for a rock track so that it peaks at -30dBfs, because it's probably going to be the loudest thing in the mix when you're done, and you'll have to fight to get it there without the rest of the kit coming with it.

 

These are extreme examples to illustrate the point, but you see what I mean. Record it how you want to hear it, and the battle is already mostly over. Do this, and pay attention to the rules David has outlined AFA peak levels, and you'll be fine.

 

Hope that's a bit more helpful.

[darkecho]

If you're clipping the plug-ins that means you're clipping the A/D converter, and that's what you should be worrying about. Furthermore you can't clip the input of a 32 bit plug-in floating point (All of Logic's plug-ins and most 3rd party plug-ins are 32 bit floating point).

 

If you're using any 24 bit fixed 3rd party plug-ins, the only way to know for sure you're not clipping it is to watch the input level (with the plug-in's own level meter, or by inserting a level meter before the plug-in), and possibly turn down the gain before that plug-in.

 

But if you follow common digital recording practice (leave yourself 6dB of headroom or more) and don't go crazy with your gain staging (don't start inserting plug-ins that add 6 dB of gain or more) you should never have to worry about any of it.

 

I am using all soft-synths at the moment.. so you said -6dBFS of headroom. So I need to set my faders to unity, and then adjust the softsynth/virtual instruments output level to peak no higher than -6dBFS?

 

that would then give me a good healthy level and some headroom for peaks, I can then adjust my faders during mixing and bring things down to maintain a Master Output level of -6dBFS?

[marcel72]

If you're clipping the plug-ins that means you're clipping the A/D converter, and that's what you should be worrying about. Furthermore you can't clip the input of a 32 bit plug-in floating point (All of Logic's plug-ins and most 3rd party plug-ins are 32 bit floating point).

 

If you're using any 24 bit fixed 3rd party plug-ins, the only way to know for sure you're not clipping it is to watch the input level (with the plug-in's own level meter, or by inserting a level meter before the plug-in), and possibly turn down the gain before that plug-in.

 

But if you follow common digital recording practice (leave yourself 6dB of headroom or more) and don't go crazy with your gain staging (don't start inserting plug-ins that add 6 dB of gain or more) you should never have to worry about any of it.

 

I am using all soft-synths at the moment.. so you said -6dBFS of headroom. So I need to set my faders to unity, and then adjust the softsynth/virtual instruments output level to peak no higher than -6dBFS?

 

that would then give me a good healthy level and some headroom for peaks, I can then adjust my faders during mixing and bring things down to maintain a Master Output level of -6dBFS?

 

Try setting your faders at unity, and then adjusting your SI levels so that the song basically sounds how you want it to, with nothing peaking higher than -6dBfs. Then when you're tweaking, your tweaks will be relatively (<6dB) small. IME, things will come together much more quickly this way. Mixing is often about eliminating variables...

[darkecho]

SI - Signal input?

 

So adjust the output of my softsynths etc to peak no higher than -6dbfs and roughly the volume I will want them to be.

 

is it better to adjust the level of an instrument by changing the output level of the synth rather than the instruments fader?

[David Nahmani]

is it better to adjust the level of an instrument by changing the output level of the synth rather than the instruments fader?

Not sure what you mean by "the instruments fader", but if you mean the fader on the channel strip in Logic, it does not control the level of the recording. Only a hardware analog gain knob placed BEFORE the A/D converter can adjust the level of a recording. You usually use the gain knob on your audio interface, or preamp.

[marcel72]

I meant Software Instrument, and I assumed that since you made a distinction between the level of the instrument and the level of the channel, that you were dealing with instruments that had some sort of gain 'knob' prior to the channel strip fader in Logic, no? This would be where I would try to establish balance, initially.

 

FWIW, 'get the level right as close as possible to the source' is kind of the engineering method that I was taught... But inside Logic, with its massive dynamic range, fantastic SNR, and floating point mixer, all this stuff is pretty much irrelevant. You can just mix how you want, and adjust the output fader to keep your levels under control if you so wish.

 

There are only 2 places that levels really matter:

 

At the input A/D

At the output D/A

 

The rest is really up to you.

[darkecho]

David, yes I was referring to the channel fader. And yes, I understand that the only way to adjust the level coming IN is to adjust the source (IE, mic distance, or preamp gain, etc) In the case of virtual instruments, the virtual instrument should have a gain knob in it. that is what I would refer to as the pre-fader level I guess

 

Marcel- yes I have gain knobs in my Virtual instruments. So acheiving the balance pre-fader is the important part..

 

what do you suggest as a process to "balance" these signals?

 

I just want to clear up what is the optimal level to "start" with (before I start mixing with the channel faders) to avoid internal plug clipping, intersample peaks, etc...