-18dBFS = 0dBV, Always?

[hduncan]

Also you mentioned that a vu meter is an averager versus the digital which measures peaks. Might I assume that this is because analogue's optimal level of 0dBVU affords us the headroom for the peaks that digital can't allow to ocurr over 0dBFS (for consequence of clipping)

so in the end, the -18dBFS = 0dBVU is saying that most analogue equipment can handle +18dB over its optimal level of 0dBVU that the DAW feeds it. Because you wouldn't want to directly compare the ceilings if both meterings on their respective platforms because they behave differently as the audio signal approaches the ceiling. So the -18dBFS is a good compromise to achieve an optimal level that transltes to both platforms

[A27Hull]

Originally Posted by peat

try this in PT

I did this with PT and have some observations to share.

I wanted to calibrate my PSX-100, and the only function generator I had was in PT.

I chose a 0VU=+4dbu=1.228 Vrms standard, and attempted to calibrate the ADDA to
-16dbfs.

I set up a session in PT with an aux track and PT function generator plugin, took a Vrms meter and measured the output (analog 1) of my 002R.

I had to tweak the plug a good bit to find a frequency to output relationsip to get 1.228 Vrms at the analog 1 output.

Long story short, I found that a 980HZ sine, -13.9 level set within the plugin, with aux fader set to null/unity, gave me a 1.228Vrms/+4dBu/0VU at analog 1.

I wouldn't assume that the level I used in my system should work form everyone else. , though. But my findings do support the line of thought that suggests the actual levels in and out of PT are misleadingly presented in the software GUI.


AfaraWayland: DIY: Calibrating AD/DA, Part Three

[Farview]

Originally Posted by DarkEcho

Also you mentioned that a vu meter is an averager versus the digital which measures peaks. Might I assume that this is because analogue's optimal level of 0dBVU affords us the headroom for the peaks that digital can't allow to ocurr over 0dBFS (for consequence of clipping)

Yes

Originally Posted by DarkEcho

so in the end, the -18dBFS = 0dBVU is saying that most analogue equipment can handle +18dB over its optimal level of 0dBVU that the DAW feeds it.

Analogue equipment can handle transients 18db above its optimal level. It doesn't do as well when the bulk of the signal is too far above line level.

Originally Posted by DarkEcho

Because you wouldn't want to directly compare the ceilings if both meterings on their respective platforms because they behave differently as the audio signal approaches the ceiling. So the -18dBFS is a good compromise to achieve an optimal level that transltes to both platforms

Yes. The other thing to take into account is that analogue equipment doesn't stay completely clean up to the point that it starts clipping. (digital does) Analogue gradually gets more distorted as the level goes past optimum until it finally clips.

[slipstream]

Originally Posted by A27Hull

I did this with PT and have some observations to share.

I wanted to calibrate my PSX-100, and the only function generator I had was in PT.

I chose a 0VU=+4dbu=1.228 Vrms standard, and attempted to calibrate the ADDA to
-16dbfs.

I set up a session in PT with an aux track and PT function generator plugin, took a Vrms meter and measured the output (analog 1) of my 002R.

I had to tweak the plug a good bit to find a frequency to output relationsip to get 1.228 Vrms at the analog 1 output.

Long story short, I found that a 980HZ sine, -13.9 level set within the plugin, with aux fader set to null/unity, gave me a 1.228Vrms/+4dBu/0VU at analog 1.

I wouldn't assume that the level I used in my system should work form everyone else. , though. But my findings do support the line of thought that suggests the actual levels in and out of PT are misleadingly presented in the software GUI.


AfaraWayland: DIY: Calibrating AD/DA, Part Three

At least 980HZ is really close to 1K tone.

[cjogo]

see below.

[cjogo]

We come off our DAW @ -12 --staying digital to our DEQ 2496 for "mastering"
and then straight to a Masterlink .

[hduncan]

So I am using the apogee duet,

in order to find out what it's sweet spot is..

A) should I consult the manual for either a specific number or a dynamic range inwhich I can infer the optimal level?

B) just make sure that the built in meters on the duet itself dont clip

C) or listen intently for something my ears arent trained to listen for ?


And again, because the only thing attached to the analogue side of my duets converters, the only reason I would care if the analogue is clipping (if the digital isnt) would be so I can hear an unclipped signal going to my monitors...

otherwise, there is no reason for me to care about the analogue side of my converters being completely sealed off ITB, right?

[Farview]

Originally Posted by DarkEcho

So I am using the apogee duet,

in order to find out what it's sweet spot is..

A) should I consult the manual for either a specific number or a dynamic range inwhich I can infer the optimal level?

B) just make sure that the built in meters on the duet itself dont clip

C) or listen intently for something my ears arent trained to listen for ?


And again, because the only thing attached to the analogue side of my duets converters, the only reason I would care if the analogue is clipping (if the digital isnt) would be so I can hear an unclipped signal going to my monitors...

otherwise, there is no reason for me to care about the analogue side of my converters being completely sealed off ITB, right?

You really don't have to worry about it.

[Bob Olhsson]

Originally Posted by DarkEcho

in order to find out what it's sweet spot is..


C) or listen intently for something my ears arent trained to listen for ?

This is where a 400 to 1000 Hz. test tone makes any problems obvious. Ideally you want to try and keep program material peak levels 6-10 dB. lower than where you hear the tone clipping.

[Farview]

To elaborate a little on my last comment...

You shouldn't have to worry too much because you have such a closed system. I doubt that your converters are capable of clipping the input of your monitors without you really deserving it.

With that said, you really should find out what line level is for your converters and calibrate your monitors.

[heavi5ide]

Check out these articles on Bob Katz's website. Good stuff. Answers a lot of these questions in depth. I read this all awhile ago in his book Mastering Audio (good read!) so I can't remember exactly what he gets into, but he's got a lot of great articles on his site.

Digital Domain - Level Practices (Part 1)

Digital Domain - Level Practices (Part 2) (Includes the K-System)

Nick

[hduncan]

Originally Posted by Farview

To elaborate a little on my last comment...

You shouldn't have to worry too much because you have such a closed system. I doubt that your converters are capable of clipping the input of your monitors without you really deserving it.

With that said, you really should find out what line level is for your converters and calibrate your monitors.

I was doing so well, and you lost me here..

if the analogue side clips as the signal comes out of the DAW, through the D to the A and to the monitors, then the monitors will receive a clipped signal and thus, will have no choice but to reproduce it, right?

and the second part... the only thing I can change on the back of my monitors is the flat-head screw for volume.. is that what you are talking about? I have no idea how this calibration process would work...

[Farview]

Originally Posted by DarkEcho

if the analogue side clips as the signal comes out of the DAW, through the D to the A and to the monitors, then the monitors will receive a clipped signal and thus, will have no choice but to reproduce it, right?

True, but in actual practice, it doesn't really happen. Since it is a DA converter, the designers know exactly what the highest signal level coming out of it will be (0dbFS) and will design the analog side of the circuit accordingly.

The place where everyone has trouble with gain staging is on the way in. That's because there are so many variables on that side. These variables don't exist between your DA and monitors.

Originally Posted by DarkEcho

and the second part... the only thing I can change on the back of my monitors is the flat-head screw for volume.. is that what you are talking about? I have no idea how this calibration process would work...

You would still have to find out what digital level in the DAW is line level on the analog side. Then you would send a test tone at that level out of the DA to your monitors. With an SPL meter, set line level to be 80 or 85dbspl.

That way, you will have a reference to how loud your mix really is and you will start blowing your ears out before you start clipping the inputs of the monitors.



The important part of this discussion that might be getting lost is: Line level is king in the analog world. Everything after the mic preamp should be recieving and outputing line level. That is proper gain staging. That is the important point.

All converters will be calibrated to 'line level'= 'something' dbfs. Line level is 0dbVU

It's up to you to find or set the calibration so that you know where you are supposed to be sitting on the meters in the box.

[hduncan]

Originally Posted by Farview

All converters will be calibrated to 'line level'= 'something' dbfs. Line level is 0dbVU

It's up to you to find or set the calibration so that you know where you are supposed to be sitting on the meters in the box.

Thanks for all of the help thus far..

I am still unsure of how to figure out what dbfs = line level for me..

I have the duet, I've been checking the specs but cant find anything related to its' dynamic range. is this where I use the test tone until it clips and then back it off a certain amount of decibels?

Once I find out the mystery dbfs value, THAT value is what I should have my mixes peak at (in the master output)?

[Farview]

Originally Posted by DarkEcho

Thanks for all of the help thus far..

I am still unsure of how to figure out what dbfs = line level for me..

I have the duet, I've been checking the specs but cant find anything related to its' dynamic range. is this where I use the test tone until it clips and then back it off a certain amount of decibels?

Once I find out the mystery dbfs value, THAT value is what I should have my mixes peak at (in the master output)?

All you need to do is get a test tone and a multi-meter.

Play the tone out of the converters and adjust the level until the meter reads 1.228 volts. (if it is a +4 unit) Then you just look at the meter in the DAW to see what it says when you are pushing a line level signal. Line level= 0dbVU = 1.228 volts. (unless it is a unit that is using -10 line level, then you should be looking for 0.3162 volts, if memory serves....)

[hduncan]

How do I tell between +4 or -10?

[Farview]

Originally Posted by DarkEcho

How do I tell between +4 or -10?

It will say in the specs. It will be called 'operating level' or something similar.

[hduncan]

Excellent, thanks!!

I found the specs of the duet,

Two channels of professional-quality 24-bit/96kHz audio input and output
Two unbalanced -10 dBV line outputs for powered speakers

Given that the outputs to speakers (which would be the analogue side) says -10, is that the magic number?

I also heard Max (apogee representative) say that the Duet's dynamic range is close to 20 bits, and the same as the Apogee ensemble, which is 114db.

Do I need the dynamic range numbers in order to figure out what my DAW should be RMSing, or just use your test tone method?

Thanks

[Farview]

Originally Posted by DarkEcho

Two channels of professional-quality 24-bit/96kHz audio input and output
Two unbalanced -10 dBV line outputs for powered speakers

Given that the outputs to speakers (which would be the analogue side) says -10, is that the magic number?

That means that line level for that unit will be 0.3162 volts. It also mean that if your monitors are +4, you are going to have a hard time driving them unless there is a way to compensate. (like a -10/+4 switch)

Originally Posted by DarkEcho

I also heard Max (apogee representative) say that the Duet's dynamic range is close to 20 bits, and the same as the Apogee ensemble, which is 114db.

Do I need the dynamic range numbers in order to figure out what my DAW should be RMSing, or just use your test tone method?[/quote]No, the dynamic range is the difference between the noise floor and the loudest signal that it can handle. You need to find the maximum output.

for example: If the max output is +8dbV and we know that line level is -10dbV, then we know that it has 18db of headroom. And, therefore, line level = -18dbFS.

You have to watch which db scale all these numbers you are collecting belong to. dbV, dbFS, and dbu are all different scales and don't relate directly to each other.

[hduncan]

I spoke to apogee tech support and he couldn't really give me an answer as far as the max output. He kept saying its fixed at -10.. but at one point he did say the numbers (-16 dbfs)

So I dont know if that means that the max output level= +6dbV... its all very confusing. I dont know why I can't find the simple max output value of this device. seems like it might be a semi-important value..

I guess I'll just shoot to for a master output RMS level of -18dbfs?

[Farview]

Originally Posted by DarkEcho

I spoke to apogee tech support and he couldn't really give me an answer as far as the max output. He kept saying its fixed at -10.. but at one point he did say the numbers (-16 dbfs)

So I dont know if that means that the max output level= +6dbV... its all very confusing. I dont know why I can't find the simple max output value of this device. seems like it might be a semi-important value..

I guess I'll just shoot to for a master output RMS level of -18dbfs?

That'll be the safest bet. I'm suprised that there are no real specs on this thing (I just looked it up). Of course I'm not suprised that the support guy doesn't know any more than the manual tells you.

[psycho_monkey]

Originally Posted by A27Hull

I did this with PT and have some observations to share.

I wanted to calibrate my PSX-100, and the only function generator I had was in PT.

I chose a 0VU=+4dbu=1.228 Vrms standard, and attempted to calibrate the ADDA to
-16dbfs.

I set up a session in PT with an aux track and PT function generator plugin, took a Vrms meter and measured the output (analog 1) of my 002R.

I had to tweak the plug a good bit to find a frequency to output relationsip to get 1.228 Vrms at the analog 1 output.

Long story short, I found that a 980HZ sine, -13.9 level set within the plugin, with aux fader set to null/unity, gave me a 1.228Vrms/+4dBu/0VU at analog 1.

I wouldn't assume that the level I used in my system should work form everyone else. , though. But my findings do support the line of thought that suggests the actual levels in and out of PT are misleadingly presented in the software GUI.

If I understand that correctly, you're measuring with a voltmeter, rather than feeding into a desk (with +4 inputs).

The 002s are designed to output at -18dBFS=0VU. The above shows that you get 0VU from -13.9 (or -17.9 when you take into account the input level of a mixing desk being a nominal +4, which your voltmeter doesn't take into account). Pretty good accuracy from a "semi-pro" interface.

[hduncan]

Alright. according to some other threads (primarily the Qs for Paul Frindle thread) I am going to keep my levels peaking at -6dBFS in the digital domain to avoid intersample peaks and what not, but this is just for digital only precautions.

if I were to go OTB with an FX box, this is where I would want to know what to calibrate my dBFS scale to Line for, right? Since the second tech I spoke to said that the Duet's converters are calibrated for -16dBFS, I should output the channel heading for the outboard FX unit at -16dBFS RMS because -16dB in Logic's Full Scale equates to line level coming out of the analogue side of the Duet's converter, and I would want to send line level to the FX unit for the optimal performance because it is in the analogue domain.

Is this all good? =) Thanks everyone!

[hduncan]

I just wanted to revive this thread for a moment incase some Apogee Ensemble/Duet users were looking for help.

I had the Duet and now have the Ensemble, I spoke with Jay at Apogee Tech Support and he said that the Ensemble has a fixed calibration of -16dBFS=Line Level.

So in the grand scheme of things, I believe this means that, if you want to send something out of the Ensemble to some outboard gear, you should get the signal that you are outputing to RMS around -16dBFS. That way the outboard gear will be receiving a line level signal as they expect.

Also, I believe this means that, when you have a sound coming into a channel in the DAW, it should RMS at -16dBFS. So you'd need to set your fader to 0.0/Unity, then adjust the input of either your softsynth/analogue gear coming into the Ensemble so that it's RMS (average) signal is at -16dBFS on the meters.

This way, any plugins you add, especially if they model analogue gear, will emulate properly. If you hit the plugin too hard, it would be like THINKING you are sending line level to some analogue gear, but you are really hitting it too hard and it is saturating.

Can someone confirm so I can graduate from Gain-Staging 101?


P.S. After you have set your channel to RMS at -16dBFS on the way in with the Fader at Unity, and you have applied all of your gain-affecting plugins and processors, you can then adjust the fader to get the signal to peak no higher than -6dBFS at the master meter (to avoid intersample peaks and keep yourself from clipping).

[colinmiller]

It simply means that if your analog gear is sitting at 0VU, then it will be sitting around -16dBFS in your DAW.

[Farview]

Originally Posted by DarkEcho

Can someone confirm so I can graduate from Gain-Staging 101?

Well, it's close enough to keep you out of trouble.

However, it seems as if you are taking the numbers too seriously and absolutely. Line level is a target, a little below or above is not the end of the world and isn't a big deal.


As for your P.S., you adjust the fader so that instrument is the proper volume in the mix. The mix is what ultimately matters. Proper gain staging is just using your equipment properly.