Pro tools calibration reference level

Home The Forums Music Computers DAW Talk Avid Pro Tools Dumb question: How can I see actual DB levels in pro tools? Not just meter

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So I've always been a logic guy, and I recently moved over to an HD system with an aurora-16 and a bunch of new pres. Where in PTHD can I get it to show me actual levels, so I can ensure I'm around ~-10 range when coming in. not just that yellow-orange area on the meter before I clip it .

I need to make sure all my gain staging is setup properly.

You should have seen how long it took me last night to find where I could set the tempo for the session!

An easy way with whats already there:

Create an Aux track and insert a signal generator.
Set the output to -10, or whatever you prefer.
Mute it and use it as a reference for your input meters.

It might not be perfect but it works.

Soon enough you'll know from memory
how high up the meter your "mark" is.

So theres nothing within protools to handle this? I considered using a 3rd party meter, but figured there must be SOMETHING within the app itself. I need to figure out what 0 on my vu meters on my pres translate to within protools. I haven't had to gain stage before.

I suppose I'll download that plug now.

An easy way with whats already there:

Create an Aux track and insert a signal generator.
Set the output to -10, or whatever you prefer.
Mute it and use it as a reference for your input meters.

It might not be perfect but it works.

Soon enough you'll know from memory
how high up the meter your "mark" is.

Could you not send a sine wave out to your pres, bring it back in
on an audio track, and Apple+click the box under the meter
to show the peak value?

Adjust your pre till the peak value in PT reads whatever you
are calibrating to?

Maybe this is wrong. Im sure someone who knows better
will chime in soon enough.

I do read manuals, and if I understand things properly, +18db should be 0dbfs within protools for the interface (lynx) I'm using. But last night when my ADL600 was barely hitting 0vu I was clipping. Maybe I don't understand all the relations well enough. Perhaps I had the ADL in a wrong VU mode! I don't know, but it was odd.

I do read manuals, and if I understand things properly, +18db should be 0dbfs within protools for the interface (lynx) I'm using. But last night when my ADL600 was barely hitting 0vu I was clipping. Maybe I don't understand all the relations well enough. Perhaps I had the ADL in a wrong VU mode! I don't know, but it was odd.

LOL. -18dBFS = 0VU is AES standard for studios. FS means Full Scale so you can't go further than 0. VU and FS are pretty different scales from two very different generations of recording gear.

VU is based around average volume. Kind of like RMS, so a snare peaking at 0VU will have transients way, way louder.

Get on Wiki and have a read about audio volume standards, but a crib sheet would suggest you don't allow your highest transient peaks (measured on normal DAW Peak meters) to go over -6 dBFS.

I see. I will read up more -- I know that these are different scales and I understand full scale within the DAW.

Not going over -6bdfs is what I aim for, and is why I was asking about metering in the first place.

That's not true. It is certainly possible to have >0 dBFS values. Just not in a 24 bit fixed point word. Modern DAW's all allow (and can mix together without internally clipping) values above 0 dBFS.

I see. I will read up more -- I know that these are different scales and I understand full scale within the DAW.

Not going over -6bdfs is what I aim for, and is why I was asking about metering in the first place.

I think your misunderstanding stems from not annotating what dB scale you are talking about.

Saying 18 dB = 0 dBFS really doesn't mean anything.

18 dBSPL would be something very quiet
+18 dBu would be a fairly large voltage in terms of professional audio equipment.

Always always always define what dB scale your number references against.

In terms of setting levels within a DAW, it gets a little tedious to actively measure the peak value while tracking. just eyeball it. The more you do it the better you get at it.

Always remember that in the digital domain setting your levels "too low" is 1000x better than too high.

That's not true. It is certainly possible to have >0 dBFS values. Just not in a 24 bit fixed point word. Modern DAW's all allow (and can mix together without internally clipping) values above 0 dBFS.

Not in a digital system, 0 dBFS is assigned to the maximum possible level of that particular system, and all measurements are given as a negative number in dB from 0dBFS. Certainly as far as metering is concerned.

If you are in PT8 the numbers are on each track in the mixer panel. I think the left is dbfs and right is vu.

That's not true. It is certainly possible to have >0 dBFS values. Just not in a 24 bit fixed point word. Modern DAW's all allow (and can mix together without internally clipping) values above 0 dBFS.

Maybe we should stick 3 or 4 dB on the PT meters to account for the interpolation of overs? But we colour them red yeah?

Not in a digital system, 0 dBFS is assigned to the maximum possible level of that particular system, and all measurements are given as a negative number in dB from 0dBFS. Certainly as far as metering is concerned.

So are you saying that it is not possible to have values above 0 dBFS in reference to metering in a DAW?

Its fairly easy to see that that is not true.

There are several plugins that can meter positive dBFS values. Dynamic Spectrum Mapper is one of them.

Perhaps we are just misunderstanding each other.

I can mix together 20 phase coherent sine waves, each with an amplitude of -.1 dBFS.

This results in a value that is much higher than 0 dBFS.

I can then route that value through the software and even through some plugins and recover that value to something below 0 dBFS without actually clipping the mix buss of the software.

Maybe we should stick 3 or 4 dB on the PT meters to account for the interpolation of overs? But we colour them red yeah?

**** that. if you are recording or mixing anywhere NEAR 0 dBFS then you deserve what you get.

I believe the metering is there to tell you that the audio HAS or WILL clip when reduced back to 24 bits fixed point and outputted or inputted through a converter.

So are you saying that it is not possible to have values above 0 dBFS in reference to metering in a DAW?

Its fairly easy to see that that is not true.

There are several plugins that can meter positive dBFS values. Dynamic Spectrum Mapper is one of them.

Perhaps we are just misunderstanding each other.

I can mix together 20 phase coherent sine waves, each with an amplitude of -.1 dBFS.

This results in a value that is much higher than 0 dBFS.

I can then route that value through the software and even through some plugins and recover that value to something below 0 dBFS without actually clipping the mix buss of the software.

PT HD isn't floating point (at least it wasn't when I was using it. may have changed) so the channels are limited at 0dbfs when manipulating audio and will clip if you go over 0dbfs. in contrast, Nuendo or Cubase allows you to overshoot 0dbfs and not clip out as long as you're not clipping out a fixed point plug or going over 0dbfs as you hit the converter output.

As to the OP, if all you just want to be sure you're not going over -6dbfs, you should be able to control-click the meter and it will show peak values.

PT HD isn't floating point (at least it wasn't when I was using it. may have changed) so the channels are limited at 0dbfs when manipulating audio and will clip if you go over 0dbfs. in contrast, Nuendo or Cubase allows you to overshoot 0dbfs and not clip out as long as you're not clipping out a fixed point plug or going over 0dbfs as you hit the converter output.

As to the OP, if all you just want to be sure you're not going over -6dbfs, you should be able to control-click the meter and it will show peak values.

And we have a winrar.

If you are in PT8 the numbers are on each track in the mixer panel. I think the left is dbfs and right is vu.

Which you will only see if you unselect "narrow mix window" or expand a particular fader control. Once you get the hang of it you'll do it 85% by eye and use the numbers less frequently.

PT HD isn't floating point (at least it wasn't when I was using it. may have changed) so the channels are limited at 0dbfs when manipulating audio and will clip if you go over 0dbfs. in contrast, Nuendo or Cubase allows you to overshoot 0dbfs and not clip out as long as you're not clipping out a fixed point plug or going over 0dbfs as you hit the converter output.

As to the OP, if all you just want to be sure you're not going over -6dbfs, you should be able to control-click the meter and it will show peak values.

Thats not true either.

The PTHD has a 48 bit fixed point mixer which provides similar headroom (and footroom!) above 0 dBFS to a 32-bit floating point mixer.

Both the native solution of floating and the hardware solution of fixed point have advantages and disadvantages.

The problem with HD is when you start mixing TDM and RTAS plugins. then it really gets fun! :-)

Thats not true either.

The PTHD has a 48 bit fixed point mixer which provides similar headroom (and footroom!) above 0 dBFS to a 32-bit floating point mixer.

Both the native solution of floating and the hardware solution of fixed point have advantages and disadvantages.

The problem with HD is when you start mixing TDM and RTAS plugins. then it really gets fun! :-)

All fine and good. but this has nothing to do with what I said. I wasn't debating the merits of fixed vs. float, nor claiming one had better or worse headroom.

I was only making the comment that in PT (at least when I was using it a couple years ago), the channels will clip at 0dbfs when mixing. So the channels have a hard stop.

You could ramp gain past the legal point with a plugin chain, then ramp back down so you're under 0dbfs on the PT meter without the reds lighting up. but I'm not entirely sure what you'd gain by doing this.

Thats not true either.

The PTHD has a 48 bit fixed point mixer which provides similar headroom (and footroom!) above 0 dBFS to a 32-bit floating point mixer.

Both the native solution of floating and the hardware solution of fixed point have advantages and disadvantages.

The problem with HD is when you start mixing TDM and RTAS plugins. then it really gets fun! :-)

The thing with that is you can't describe any of the calculations (48bit floating whatever) with a dB value. The headroom of a floating point system is so ridiculously high as to be considered limitless.
Sound goes in > Gets fiddled around with > Sound comes out.

It can never go in or come out over 0dBFS.
Maybe your converters will try and represent the overs by guessing, but 0dBFS is the ceiling when talking about sound you can actually hear. (My Favourite kind).

All fine and good. but this has nothing to do with what I said. I wasn't debating the merits of fixed vs. float, nor claiming one had better or worse headroom.

I was only making the comment that in PT (at least when I was using it a couple years ago), the channels will clip at 0dbfs when mixing. So the channels have a hard stop.

You could ramp gain past the legal point with a plugin chain, then ramp back down so you're under 0dbfs on the PT meter without the reds lighting up. but I'm not entirely sure what you'd gain by doing this.

No the channels do not have a hard stop. Take a -.1 dBFS sine wave. turn the fader up to +12. The channel is now outputting a sine wave at +11.9 dBFS. (In fact, I believe you can do this with >128 phase coherent -.1 dBFS sine waves ALL with their channel faders at +12 and STILL not see the upper limits of the mix buss in either fixed or floating point systems. )

Then use a masterfader to recover the sine wave to below 0 dBFS before outputting to a converter.

Remember that the only reason I bring this up is because MarkRB said that there was no values higher than 0 dBFS.

You should NOT be anywhere near the maximum of the system for a variety of OTHER reasons.

The thing with that is you can't describe any of the calculations (48bit floating whatever) with a dB value. The headroom of a floating point system is so ridiculously high as to be considered limitless.
Sound goes in > Gets fiddled around with > Sound comes out.

It can never go in or come out over 0dBFS.
Maybe your converters will try and represent the overs by guessing, but 0dBFS is the ceiling when talking about sound you can actually hear. (My Favourite kind).

Yes, of course. the 24 bit fixed point world of the A/D or D/A converter will not pass values above 0 dBFS. because it has run out of numbers! heh

But you CAN describe calculations above the 0 dBFS value and there are MANY plugins that do make calculations AND display the metering information as positive dBFS values.

Again, I am not arguing the merits of mixing above 0 dBFS. because there aren't any! hehhehheh

Just merely stating that it is indeed possible to have positive dBFS numbers. thats all. no more. no less! :-)